Re: [SR-Users] Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP
Indeed, which works for simple demos and fits on a single slide - the whole purpose of that presentation. If someone is building a production system they really need to understand the various use-cases they will see and write their Kamailio configuration properly. Regards, Peter On 6 April 2014 19:58, Juha Heinanen j...@tutpro.com wrote: Olli Heiskanen writes: Thanks, I'll look into the rtpengine, had a busy weekend but next week I'll have better time. what comes to peter's slideshare failure_route example, i think it only works in very simple unrealistic scenario when there is no forking or serial routing. also, its nathelper handling is unnecessary when websocket sip ua, such as jssip, supports gruu. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Acision ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Regarding Websocket module
RHEL 6.x, CentOS 6.x, and Fedora all have this. It isn't present in RHEL/CentOS 5.x Regards, Peter On 2 April 2014 09:18, Juha Heinanen j...@tutpro.com wrote: Premchandiran writes: Not able to find libunistring-dev rpm for Linux 2.6.18-274.el5. almost 4 hours I am searching for the rpm. switch to debian, -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Acision ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Does tls/wss actually work or What is required for tls/wss
Hello,
Probably a silly question, but does your xhttp event_route go on to
actually handle the WebSockets handshake?
There is an example websocket.cfg in the examples directory in Git. Have
you tried using this?
Peter
On 19 March 2014 16:03, jaflong jaflong jafl...@yandex.com wrote:
Ollie, Thanks for the info.
I am not aware how to test SIP/TLS can you make a suggestion of how to do
it and what is the url of the page you mention.
However I have followed this page
http://www.kamailio.org/wiki/tutorials/tls/testing-and-debugging
I can get a successful tls connection when I connect with http so I know
basic tls works.
Tested by having this in kamailio.cfg
event_route[xhttp:request] {
set_reply_close();
set_reply_no_connect();
xhttp_reply(200, OK, text/html,htmlbodyReceived HTTP
request to $hu from [$si:$sp] with protocol $proto/body/html);
xlog(L_INFO, HTTP Request Received\n);
..
Going to https://10.1.2.3:6443 gives this
Received HTTP request to / from [10.1.1.1:58179] with protocol tls
19.03.2014, 19:50, Olle E. Johansson o...@edvina.net:
On 19 Mar 2014, at 16:46, jaflong jaflong jafl...@yandex.com wrote:
Hi,
What are the requirements for connecting with tls/wss.
I have not come across any information or example for this.
My config is working when the client uses ws. However if I change this
to use wss, (this is it only paramter I change) it does not work.
I understand Kamailio does not support DTLS, I set the jssip client
DtlsSrtpKeyAgreement to false to disable this, I also set the tls option to
not require or verify certicficates and it still does not work.
Kamailio has nothing to do with DTLS - it's in the media layer, not in
the signalling.
What if other considaerstion do I need to check?
Check if normal SIP/TLS works and if you can connect with a web browser.
There is a TLS debugging page on the Kamailio wiki with a lot of helpful
tips and tricks. We might want to add WSS to that page.
/O
thanks
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Re: [SR-Users] Using Kamailio as a websocket LB
No that is not currently possible. Kamailio is not an HTTP load-balancer. If you want a WebSocket load-balancer I suggest you look at NGiNX. Regards, Peter On 21 February 2014 10:13, Luis Silva luisfilsi...@gmail.com wrote: Hi guys, Is it possible to use Kamailio as a Websocket LoadBalancer (transparent to the Websocket content, rather it's SIP or other protocol)? Many thanks, Luís ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Using the auth_ephemeral module
Hello, You have to write the web-service yourself. The IETF draft referenced in the module documentation explains how the web-service should construct the credentials - the coding for this is trivial. The mechanism the web-service uses to authenticate the user in the first place (and decide whether to issue credentials or not) will vary from application to application and is entirely up to you. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd On 21 Nov 2013, at 03:09, Hemanshu Vadehra hemansh...@directi.com wrote: I'm involved in setting up a Kamailio instance and was hoping to make use of the auth_ephemeral module for authentication. But the module documentation doesn't quite make clear how exactly the module is to be employed or the web service set up. Does anyone have a working example? Regards, Hemanshu Vadehra hemansh...@directi.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Steps to next major release v4.1.0
No. There has been no development done on this module for several months. I can't speak for any of the other developers, but I don't have any plans to work on this unless there is a specific demand for it from a customer. Regards, Peter On 12 November 2013 21:40, mohammed alyaseen alyasee...@yahoo.com wrote: Hello, Is the new release 4.1.0 with completed MSRP module? thanks, Medo Daniel-Constantin Mierla mico...@gmail.com schrieb am 21:39 Dienstag, 12.November 2013: Hello, I am planning to make the branch for 4.1 on Thursday. Afterwards, the master can get code for new features and fixes will have to be backported from master to 4.1 branch (and older, when it is the case). If anyone is considering alternatives, then reply here with the options. Cheers, Daniel On 11/7/13 10:01 AM, Daniel-Constantin Mierla wrote: Hello, it's now one month since we froze the development for release of v4.1.0. No critical issues in on the table for this particular version, so perhaps next week is time to create a dedicated branch for it, to be named in GIT as 4.1, and open development for v4.2. If all goes fine for one more week or so, then we can do the release. Don't forget to add any open issues you are aware of to the tracker so we can solve them in time. If any of devs or community members have spare time, adding ids to the xml docbook files (for modules) will help getting a better alpha-numeric index for parameters and functions -- see more at: - http://www.kamailio.org/wiki/devel/module-docbook-readme#section_ids Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Steps to next major release v4.1.0
The missing features are: - Ability to handle indeterminately large incoming MSRP SEND requests and use TCP windowing for MSRP congestion control (so not using the fixed size TCP receive buffers currently in Kamailio). The way to do this is for Kamailio to chunk the large incoming MSRP messages as it receives them and send smaller outgoing messages ones instead of attempting to receive the entire MSRP SEND request into a single buffer. - Failure delivery reports Full details of what is required is documented here: http://www.kamailio.org/wiki/devel/completing_msrp Both of these features are critical for use with any MSRP client not specifically implemented to work-around the fact that Kamailio doesn't support these (such as the Crocodile MSRP stack). Regards, Peter On 12 November 2013 22:20, Daniel-Constantin Mierla mico...@gmail.comwrote: Mohammed, perhaps a better approach is to ask what missing feature are you looking for. IIRC, the open part was related to delivery of reports. But msrp module is ready for usage since couple of releases ago. Cheers, Daniel On 11/12/13 11:00 PM, Peter Dunkley wrote: No. There has been no development done on this module for several months. I can't speak for any of the other developers, but I don't have any plans to work on this unless there is a specific demand for it from a customer. Regards, Peter On 12 November 2013 21:40, mohammed alyaseen alyasee...@yahoo.com wrote: Hello, Is the new release 4.1.0 with completed MSRP module? thanks, Medo Daniel-Constantin Mierla mico...@gmail.com schrieb am 21:39 Dienstag, 12.November 2013: Hello, I am planning to make the branch for 4.1 on Thursday. Afterwards, the master can get code for new features and fixes will have to be backported from master to 4.1 branch (and older, when it is the case). If anyone is considering alternatives, then reply here with the options. Cheers, Daniel On 11/7/13 10:01 AM, Daniel-Constantin Mierla wrote: Hello, it's now one month since we froze the development for release of v4.1.0. No critical issues in on the table for this particular version, so perhaps next week is time to create a dedicated branch for it, to be named in GIT as 4.1, and open development for v4.2. If all goes fine for one more week or so, then we can do the release. Don't forget to add any open issues you are aware of to the tracker so we can solve them in time. If any of devs or community members have spare time, adding ids to the xml docbook files (for modules) will help getting a better alpha-numeric index for parameters and functions -- see more at: - http://www.kamailio.org/wiki/devel/module-docbook-readme#section_ids Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] XMPP- component mode
Hello, The wiki page you've referenced is up-to-date and the status of the MSRP module is that it works - with those limitations. There are SIP clients out there (for example, Blink) which support MSRP relay - I have no idea whether it works with the Kamailio MSRP relay (due to the limitations described on the wiki page) though. The crocodile-msrp stack (for in-browser use) was implemented to work-around the Kamailio MSRP limitations while still being compliment to the RFCs. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd On 22 Oct 2013, at 08:24, Wingsravi R wingsravi...@gmail.com wrote: Dear Daniel, Thank you for the reply, I even tried MSRP module too, but while i found some limitations over some features implementations in MSRP (http://www.kamailio.org/wiki/devel/completing_msrp). So can you please tel me about the current status of the MSRP relay module in kamailio ? And seems like there were no much clients that support MSRP relay extension (RFC 4976). After all the things i have tried with SIP-XMPP, as you suggested about file transfer reality with that method now i really have to back to MSRP module implementation in kamailio configuration and i will try it and back to you. Regards Nandini On Tue, Oct 22, 2013 at 12:22 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, xmpp module is only for instant messaging and parts of the sip-xmpp presence gateway. File transfer doesn't work between sip and xmpp. If you want files transfer between two sip clients, they need to implement MSRP and you have to use the msrp module from kamailio - see its readme for more details about configuring it. Cheers, Daniel On 10/19/13 8:35 AM, Wingsravi R wrote: Anybody there to help. I really want somebody to see this. please spare some time on this and give me some suggestions. Any help will meant a lot. Regards, On Wed, Oct 16, 2013 at 4:45 PM, Wingsravi R wingsravi...@gmail.com wrote: Hi Kamailio community, I am working around with Kamailio (V 4.0.3), its XMPP module- component mode and jabbed2 server, intended to get file transfer feature between two SIP clients. In a way while surfing through the blogs i got some info like this: 'In component mode, a sub domain is diverted to respective component,so you don't need users in XMPP sever'. (http://lists.sip-router.org/pipermail/sr-users/2010-August/065209.html). With this my question is: what does it mean ? Is it mean like i dont need to register xmpp clients to jabbedd2(xmpp) server ? If this is the case, How can i use my SIP users (registered to kamailio server) in jabberd2 server context ? Even XMPP module's man page doesn't give clarity about these questions. Any help will greatly appreciate. Regards, Ravi ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28; Miami, Nov 18-20, 2013 - more details about Kamailio trainings at http://www.asipto.com - ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Parameters list transformations comma and semi column conflict
Hello,
There should be no '' around the delimiter. That's why the documentation
says single character. The '' in the documentation are there because the
convention in the documentation is to surround literal examples in '' - you
can see this throughout that wiki-page.
I neither expected comma to work as a delimiter nor expected it not to
work. However, I did think there was a chance it would. I now suspect
comma is one of the few delimiters that won't work and this is probably
because the transformation parser doesn't handle them as it interprets them
as the separator for transformation arguments.
As was pointed out these transformations were originally for parsing SIP
header and URI parameters which are always delimited with semi-colon. I
extended them to take an optional delimiter so that they could be used to
parse HTTP URI parameters which are delimited by ampersand. I would never
expect to have comma delimited parameters in SIP or HTTP, so I did not
consider comma when I extended the transformation.
Regards,
Peter
On 18 October 2013 21:54, Seudin Kasumovic seudin.kasumo...@gmail.comwrote:
Hi Peter,
I tried transformation {pram.value,name['delimiter']} for 4.1.0-pre0:
$(*avp*(my_var){param.value, a, ';'})
and got parsing error:
0(32237) ERROR: pv [pv_trans.c:2386]: tr_parse_paramlist(): invalid
separator in transformation: value,a,';'}
0(32237) ERROR: core [pvapi.c:1586]: tr_lookup(): error parsing
[{param.value,a,';'}]
0(32237) ERROR: core [pvapi.c:972]: pv_parse_spec2(): bad tr in pvar
name avp
0(32237) ERROR: core [pvapi.c:998]: pv_parse_spec2(): invalid parsing in
[$(avp(my_var){param.value,a,';'})] at (4)
If set delimiter without quotes (documentation isn't clear for this), e.g.
$(*avp*(my_var){param.value, a, ;})
then no parsing complain. But, get same wrong results, for parameter
value not quoted if contains ','.
Is this expected behavior?
Regards,
Seudin
On Fri, Oct 18, 2013 at 10:41 AM, Seudin Kasumovic
seudin.kasumo...@gmail.com wrote:
Hello,
will try this feature...
thank you Peter.
On Thu, Oct 17, 2013 at 5:36 PM, Peter Dunkley
peter.dunk...@crocodilertc.net wrote:
Hello,
Parameters to SIP headers are ';' separated. ',' is used to concatenate
multiple headers onto a single line. The {param...} transformation is
intended to process SIP header parameters.
However, there is a new feature in Kamailio 4.1 (currently in a
pre-release/testing phase) that allows you to specify the delimiter value.
That may do what you require.
Please see:
http://www.kamailio.org/wiki/cookbooks/devel/transformations#paramvalue_name_delimiter
Regards,
Peter
On 17 October 2013 16:04, Seudin Kasumovic
seudin.kasumo...@gmail.comwrote:
Hi,
Transformation {param.value, param_name} returns incomplete or empty
values when parameter value contains comma (,).
See next example:
*$avp*(my_var)=a=val_a1,val_a2,val_a3;b=val_b;
in next transformations:
$(*avp*(my_var){param.value, a}) returns 'val_a1'
$(*avp*(my_var){param.value, b}) returns empty string
Seams that comma in parameter value conflicts with semi column
separator.
Is this bug or wrong documented?
Related link:
http://www.kamailio.org/wiki/cookbooks/4.0.x/transformations#paramvalue_name
--
Seudin Kasumovic
Tuzla, Bosnia
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Re: [SR-Users] how to replace sdp c line?
I've done similar things before by using a reg-ex substitution on the message body. I think I used replace_body_re() from textops. Regards, Peter On 17 October 2013 08:06, Juha Heinanen j...@tutpro.com wrote: i have tried various textops functions to replace sdp c lines with c=IN IP4 0.0.0.0 but so far all have appended the above line to the end of sdp rather than replaced existing c lines. any hints on a solution? -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] how to replace sdp c line?
I may have used replace_body_all() or replace_body_atonce() as well. On 17 October 2013 09:27, Peter Dunkley peter.dunk...@crocodilertc.netwrote: I've done similar things before by using a reg-ex substitution on the message body. I think I used replace_body_re() from textops. Regards, Peter On 17 October 2013 08:06, Juha Heinanen j...@tutpro.com wrote: i have tried various textops functions to replace sdp c lines with c=IN IP4 0.0.0.0 but so far all have appended the above line to the end of sdp rather than replaced existing c lines. any hints on a solution? -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] how to replace sdp c line?
I was using one of these functions on SDP with Git master around 3 or 4 months ago. I can't remember exactly what I did as fixes to the equipment I was connected to allowed me to take out the hack, but it worked at that point. Regards, Peter On 17 October 2013 09:53, Juha Heinanen j...@tutpro.com wrote: Peter Dunkley writes: I think I used replace_body_re() from textops. peter, replace_body_re() does not exist, but replace_body(re,txt) does. i made a test call: replace_body(c=IN IP4, c=IN IP5); and result was that original c=IN IP4 ... line is still in outgoing request, but a new line c=IN IP5 is added as the last line to the sdp. very weird. perhaps a bug in the function? -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Parameters list transformations comma and semi column conflict
Hello,
Parameters to SIP headers are ';' separated. ',' is used to concatenate
multiple headers onto a single line. The {param...} transformation is
intended to process SIP header parameters.
However, there is a new feature in Kamailio 4.1 (currently in a
pre-release/testing phase) that allows you to specify the delimiter value.
That may do what you require.
Please see:
http://www.kamailio.org/wiki/cookbooks/devel/transformations#paramvalue_name_delimiter
Regards,
Peter
On 17 October 2013 16:04, Seudin Kasumovic seudin.kasumo...@gmail.comwrote:
Hi,
Transformation {param.value, param_name} returns incomplete or empty
values when parameter value contains comma (,).
See next example:
*$avp*(my_var)=a=val_a1,val_a2,val_a3;b=val_b;
in next transformations:
$(*avp*(my_var){param.value, a}) returns 'val_a1'
$(*avp*(my_var){param.value, b}) returns empty string
Seams that comma in parameter value conflicts with semi column separator.
Is this bug or wrong documented?
Related link:
http://www.kamailio.org/wiki/cookbooks/4.0.x/transformations#paramvalue_name
--
Seudin Kasumovic
Tuzla, Bosnia
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Re: [SR-Users] pkg.stats problem, possible memory leak in websockets.
Done. The patch is in master and the 4.0 branch. Thanks, Peter On 1 October 2013 08:22, Vitaliy Aleksandrov vitalik.v...@gmail.com wrote: Hello, Thank you for the explanation. Could somebody review the patch in the attachment ? I tried to fix the problem with a growing tcpconn-refcnt for websocket connections. On 30 September 2013 17:14, Vitaliy Aleksandrov vitalik.v...@gmail.comwrote: Could you please share why nathelper aggregates both WS and WSS transports to ws and then msg_translator have to detect the type of a connection to a destination to build correct via ? modules/nathelper/nathelper.c create_rcv_uri() function : case PROTO_WS: case PROTO_WSS: proto.s = WS; proto.len = 2; break; Because when the transport is WS (WebSockets over TCP) the URI has a transport parameter like this ;transport=ws and when the transport is WSS (Secure WebSockets over TLS over TCP) the URI has a transport parameter like this ;transport=ws. In other words, the transport parameter is the same for both and you need to make the determination within Kamailio core by checking how the specified socket is actually used. -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] pkg.stats problem
On 30 September 2013 17:14, Vitaliy Aleksandrov vitalik.v...@gmail.comwrote: Could you please share why nathelper aggregates both WS and WSS transports to ws and then msg_translator have to detect the type of a connection to a destination to build correct via ? modules/nathelper/nathelper.c create_rcv_uri() function : case PROTO_WS: case PROTO_WSS: proto.s = WS; proto.len = 2; break; Because when the transport is WS (WebSockets over TCP) the URI has a transport parameter like this ;transport=ws and when the transport is WSS (Secure WebSockets over TLS over TCP) the URI has a transport parameter like this ;transport=ws. In other words, the transport parameter is the same for both and you need to make the determination within Kamailio core by checking how the specified socket is actually used. -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websockets Keep-Alive
The Kamailio websocket module sends WebSocket pongs in response to WebSocket pings from websocket clients. It can also be configured to send WebSocket pings on idle connections (and does so by default). There is no TCP level stuff here, this is all at the WebSocket layer. Regards, Peter On 26 September 2013 12:37, Juha Heinanen j...@tutpro.com wrote: Klaus Darilion writes: Question to the experts: Is keep-alive for the Websockets TCP connection automatically done by the Websockets Layer (client or server), or do I have to do it manually (nathelper pinging). since websockets uses tcp, kamailio running on linux should automatically do tcp level keepalives. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websockets Keep-Alive
tcp_keepalive=yes has an effect on the underlying TCP connection. However, simply keeping the TCP connection alive will not stop the client or server WebSocket implementation explicitly closing the connection if no WebSocket frames are received. If you are using WebSockets, and you are unsure as to the behaviour of the WebSocket client (some clients will send pings themselves and some won't - it's an implementation choice) the WebSocket server should send pings on idle connections that need to be kept open. If you want to make sure the underlying TCP connection doesn't go on you then you need to set the TCP connection parameters accordingly (the way I do it is to set the TCP connection timeout to a few seconds more than the WebSocket ping interval) - and do so for all LAN equipment in the path. A good example is that if you are using Amazon Elastic Load-Balancer to distribute WebSocket connections, idle connections will be timed-out (by the Load-Balancer) after 60 seconds - so make sure the server sends WebSocket pings more frequently than that. Regards, Peter On 26 September 2013 14:21, Juha Heinanen j...@tutpro.com wrote: Peter Dunkley writes: There is no TCP level stuff here, this is all at the WebSocket layer. Take a look at the keepalive_.* modparams for the websocket module. are you saying that websocket transport is not implemented on top of tcp? if it is then tcp_keepalive=yes core param affects also websocket transport. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websockets Keep-Alive
The Kamailio WebSocket stack will only ping on idle connections. If nathelper is sending SIP level pings on a shorter interval than the WebSocket stack is the connection won't be considered idle. The load-balancer doesn't understand the TCP traffic so it won't care as long as there is traffic. However, it would seem silly to me to use nathelper pings over the WebSocket transport (unless you already have them enabled for another transport) as it is more efficient just to let the WebSocket layer take care of it. After all, the nathelper pings from Kamailio are something of work-around that pre-dates RFC 5626. Under outbound it's quite clear that if SIP-level keep-alives are needed for a particular transport the UA should do it, not the server. Regards, Peter On 26 September 2013 14:55, Juha Heinanen j...@tutpro.com wrote: Peter Dunkley writes: A good example is that if you are using Amazon Elastic Load-Balancer to distribute WebSocket connections, idle connections will be timed-out (by the Load-Balancer) after 60 seconds - so make sure the server sends WebSocket pings more frequently than that. peter, websocket readme has this: 4.1. keepalive_mechanism (integer) The keep-alive mechanism to use for WebSocket connections. Note If nathelper is only being used for WebSocket connections then nathelper NAT pinging is not required. If nathelper is used for WebSocket connections and TCP/TLS aliasing/NAT-traversal then WebSocket keep-alives are not required. based on what you write now, is the above readme text still valid, i.e., are nat pings enough to prevent amazon load balancer from timing out websocket connections or are native websocket pings needed instead? -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Decoding HTTP URLs in event_route[xhttp:request]
Hello,
I have added a transformation to the xhttp module that breaks a URL into a
path and a querystring
- {url.path}
- {url.querystring}
I have also added an optional delimiter parameter to the {param.}
transformations.
Regards,
Peter
On 22 September 2013 14:55, Ovidiu Sas o...@voipembedded.com wrote:
You can use {s.select,index,separator} to extract the path and the
parameters into two different variables.
Or here you could create a new url transformation to break it in two:
- {url.path}
- {url.searchpath}
After that, the existing code for param transformation may be reused
(by making the separator configurable (using '' instead of ';') and
we could have a new transformation:
- {urlsearchpath.value,name}
Or maybe we can enhance the existing param transformation to pass as
an optional argument - the param delimiter:
- {param.value,name,[param_delimiter]}.
- {param.valueat,index,[param_delimiter]}
- {param.name,index,[param_delimiter]}
- {param.count,[param_delimiter]}
Regards,
Ovidiu Sas
On Sun, Sep 22, 2013 at 4:50 AM, Peter Dunkley
peter.dunk...@crocodilertc.net wrote:
Hello,
Does anyone have any ideas about this?
If not it's something I want to try and do before the freeze (any
suggestions as to how would be appreciated) as it will be a nice
finishing
touch to the WebSocket/outbound/stun/auth_ephemeral stuff I've worked on
over the last couple of releases.
Thanks,
Peter
On 19 September 2013 21:36, Peter Dunkley
peter.dunk...@crocodilertc.net
wrote:
Hello,
I was wondering if there was an easy way to decode HTTP URLs in
event_route[xhttp:request]?
For example, it would be good to be able to breakdown a URL like:
/sip?apiKey=abcdefgusername=1234567890:al...@example.com
into path/on/server (/sip in this case) and a set of parameters.
For
the parameters something like the {param.value,name} transformation for
SIP
header parameters would be ideal (which works perfectly for picking
values
out of HTTP Cookie: headers).
I noticed that there is already an {s.urldecode.param} transformation in
the PV module but I couldn't find any documentation for it in the wiki
and
looking at the code it doesn't appear to do this anyway.
Regards,
Peter
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Re: [SR-Users] Decoding HTTP URLs in event_route[xhttp:request]
Of course I'll update the wiki. I always do when I make a change like that
:-)
On 25 September 2013 16:22, Ovidiu Sas o...@voipembedded.com wrote:
Great!
Now don't forget to update the wiki:
-
http://www.kamailio.org/wiki/cookbooks/devel/transformations#parameters_list_transformations
and create the new entry for url transformations:
-
http://www.kamailio.org/wiki/cookbooks/devel/transformations#url_transformations
Regards,
Ovidiu Sas
On Wed, Sep 25, 2013 at 11:15 AM, Peter Dunkley
peter.dunk...@crocodilertc.net wrote:
Hello,
I have added a transformation to the xhttp module that breaks a URL into
a
path and a querystring
- {url.path}
- {url.querystring}
I have also added an optional delimiter parameter to the {param.}
transformations.
Regards,
Peter
On 22 September 2013 14:55, Ovidiu Sas o...@voipembedded.com wrote:
You can use {s.select,index,separator} to extract the path and the
parameters into two different variables.
Or here you could create a new url transformation to break it in two:
- {url.path}
- {url.searchpath}
After that, the existing code for param transformation may be reused
(by making the separator configurable (using '' instead of ';') and
we could have a new transformation:
- {urlsearchpath.value,name}
Or maybe we can enhance the existing param transformation to pass as
an optional argument - the param delimiter:
- {param.value,name,[param_delimiter]}.
- {param.valueat,index,[param_delimiter]}
- {param.name,index,[param_delimiter]}
- {param.count,[param_delimiter]}
Regards,
Ovidiu Sas
On Sun, Sep 22, 2013 at 4:50 AM, Peter Dunkley
peter.dunk...@crocodilertc.net wrote:
Hello,
Does anyone have any ideas about this?
If not it's something I want to try and do before the freeze (any
suggestions as to how would be appreciated) as it will be a nice
finishing
touch to the WebSocket/outbound/stun/auth_ephemeral stuff I've worked
on
over the last couple of releases.
Thanks,
Peter
On 19 September 2013 21:36, Peter Dunkley
peter.dunk...@crocodilertc.net
wrote:
Hello,
I was wondering if there was an easy way to decode HTTP URLs in
event_route[xhttp:request]?
For example, it would be good to be able to breakdown a URL like:
/sip?apiKey=abcdefgusername=1234567890:al...@example.com
into path/on/server (/sip in this case) and a set of parameters.
For
the parameters something like the {param.value,name} transformation
for
SIP
header parameters would be ideal (which works perfectly for picking
values
out of HTTP Cookie: headers).
I noticed that there is already an {s.urldecode.param} transformation
in
the PV module but I couldn't find any documentation for it in the
wiki
and
looking at the code it doesn't appear to do this anyway.
Regards,
Peter
--
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Technical Director
Crocodile RCS Ltd
--
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Re: [SR-Users] Record routing transport=tcp and examples/websocket.cfg
The above sanity check is intended to make sure that:
- SIP over TCP/TLS arriving on the ports intended for SIP over WebSocket
is correctly rejected
- SIP over TCP/TLS arriving on the port intended for MSRP is correctly
rejected
In examples/websocket.cfg with default settings:
- SIP over UDP is supported on port 5060
- SIP over TCP is supported on port 5060
- SIP over TLS is supported on port 5061
- SIP over WS is supported on port 80
- SIP over WSS is supported on port 443
- MSRP is supported over TLS on port 9000
- traffic of the wrong type/transport arriving on the wrong ports is
correctly rejected
I believe that the check you have described, and the related ones in
onreply_route, event_route[xhttp:request], and event_route[msrp:frame-in],
are not too tough do exactly what they are meant to. Whether the checks
are too tough for your exact use case depends on what that is.
Regards,
Peter
On 24 September 2013 12:30, Mikko Lehto msle...@iki.fi wrote:
Hi websocketeers, examples/websocket.cfg starts with this check:
---
request_route {
if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT)
!(proto == WS || proto == WSS)) || $Rp == MY_MSRP_PORT)
{
xlog(L_WARN, SIP request received on $Rp\n);
sl_send_reply(403, Forbidden);
exit;
}
---
My in-dialog SIP over TCP requests got 403 treatment because of this.
I believe reason was MY_WS_ADDR in route set, added by record_route().
Above sanity check seems sensible tough, so I fixed advertised route set
with call to force_send_socket() as seen on attached patch.
I wonder if my fix is the best approach.
Should the remaining other two t_relay() calls also be prepared
with force_send_socket()?
--
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Re: [SR-Users] Decoding HTTP URLs in event_route[xhttp:request]
Hello,
Does anyone have any ideas about this?
If not it's something I want to try and do before the freeze (any
suggestions as to how would be appreciated) as it will be a nice finishing
touch to the WebSocket/outbound/stun/auth_ephemeral stuff I've worked on
over the last couple of releases.
Thanks,
Peter
On 19 September 2013 21:36, Peter Dunkley peter.dunk...@crocodilertc.netwrote:
Hello,
I was wondering if there was an easy way to decode HTTP URLs in
event_route[xhttp:request]?
For example, it would be good to be able to breakdown a URL like:
/sip?apiKey=abcdefgusername=1234567890:al...@example.com
into path/on/server (/sip in this case) and a set of parameters. For
the parameters something like the {param.value,name} transformation for SIP
header parameters would be ideal (which works perfectly for picking values
out of HTTP Cookie: headers).
I noticed that there is already an {s.urldecode.param} transformation in
the PV module but I couldn't find any documentation for it in the wiki and
looking at the code it doesn't appear to do this anyway.
Regards,
Peter
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
--
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[SR-Users] Decoding HTTP URLs in event_route[xhttp:request]
Hello,
I was wondering if there was an easy way to decode HTTP URLs in
event_route[xhttp:request]?
For example, it would be good to be able to breakdown a URL like:
/sip?apiKey=abcdefgusername=1234567890:al...@example.com
into path/on/server (/sip in this case) and a set of parameters. For
the parameters something like the {param.value,name} transformation for SIP
header parameters would be ideal (which works perfectly for picking values
out of HTTP Cookie: headers).
I noticed that there is already an {s.urldecode.param} transformation in
the PV module but I couldn't find any documentation for it in the wiki and
looking at the code it doesn't appear to do this anyway.
Regards,
Peter
--
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Technical Director
Crocodile RCS Ltd
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Re: [SR-Users] Calling users on another kamailio server
Hello, It'd be better if the add_path() function could be used here. That way, if using outbound (RFC5626), the flow-token (the userinfo part of the Path-URI) would be present and there would be no need to add the ;received parameter. This would address the one issue remaining for SIP outbound on Kamailio, which is its use without an edge proxy that is separate from the registrar. Regards, Peter On 16 September 2013 14:05, Charles Chance charles.cha...@sipcentric.comwrote: Hi, This sounds like a case for sharing same database, and adding Path before saving incoming register. That way, no need to replicate register message to other servers and all subscribers use the same domain. Add path something like this before calling save(): append_hf(Path: sip:$Ri:$Rp; received=sip:$si:$sp;lr\r\n); msg_apply_changes(); Whichever server receives the incoming invite, will perform lookup and automatically route to the server which received the register. On the proxying server set $du according to received param of route header, add record-route, and then t_relay(). As Daniel said, no need to re-authenticate or perform lookup again. Regards, Charles On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 9/12/13 10:08 PM, Brian Wallen wrote: I currently have two independent kamailio servers. I'd like to set them up in a way that user1 on server1 can make a call to user2 on server2. After searching I've come up with two ways that this might be able to be done. Can someone please sanity check these or point me in the right direction? 1. Have one registrar server and convert the other server to a proxy 2. Keep them both as registrars and somehow make them each aware of the users on the other server I like 2 better because if one server went down users on the other server would still be up. The only thing is I don't know how to set the servers up to communicate with each other. the nat can create problems when a server is down - if the nat is symmetric, only the server that received the registrar can send back calls to the phone. Communication between users on two servers is as simple as using t_relay_to(proto:serverip:**port) after you do lookup(location) and no record is found. Suppose I have three or more kamailio servers. If a call comes in and lookup() returns that no record was found, how do I know which server to forward to? Is that a case in which I should replicate the database? You have to add an extra check for the case the call was coming from the other server, not to forward back to it in case of no found again. Thanks for the tip, I hadn't thought of that. Also, you should skip user authentication for calls from the other server (not do authenticate twice). Another aspect to take care is chaining rtpproxy, you have to use the flat for trusting the other server (r, iirc). You can also replicate the registration, but again, it can add troubles to the nat. Look at t_replicate() (in tm module). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013 - more details about Kamailio trainings at http://www.asipto.com - __**_ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users www.sipcentric.com Follow us on twitter @sipcentric http://twitter.com/sipcentric Sipcentric Ltd. Company registered in England Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Calling users on another kamailio server
Is there any reason it isn't possible? On 16 September 2013 14:21, Charles Chance charles.cha...@sipcentric.comwrote: Hi, Yes, you are right - and I agree, it would be better if this was possible :) Charles On 16 September 2013 14:15, Peter Dunkley peter.dunk...@crocodilertc.netwrote: Hello, It'd be better if the add_path() function could be used here. That way, if using outbound (RFC5626), the flow-token (the userinfo part of the Path-URI) would be present and there would be no need to add the ;received parameter. This would address the one issue remaining for SIP outbound on Kamailio, which is its use without an edge proxy that is separate from the registrar. Regards, Peter On 16 September 2013 14:05, Charles Chance charles.cha...@sipcentric.com wrote: Hi, This sounds like a case for sharing same database, and adding Path before saving incoming register. That way, no need to replicate register message to other servers and all subscribers use the same domain. Add path something like this before calling save(): append_hf(Path: sip:$Ri:$Rp; received=sip:$si:$sp;lr\r\n); msg_apply_changes(); Whichever server receives the incoming invite, will perform lookup and automatically route to the server which received the register. On the proxying server set $du according to received param of route header, add record-route, and then t_relay(). As Daniel said, no need to re-authenticate or perform lookup again. Regards, Charles On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 9/12/13 10:08 PM, Brian Wallen wrote: I currently have two independent kamailio servers. I'd like to set them up in a way that user1 on server1 can make a call to user2 on server2. After searching I've come up with two ways that this might be able to be done. Can someone please sanity check these or point me in the right direction? 1. Have one registrar server and convert the other server to a proxy 2. Keep them both as registrars and somehow make them each aware of the users on the other server I like 2 better because if one server went down users on the other server would still be up. The only thing is I don't know how to set the servers up to communicate with each other. the nat can create problems when a server is down - if the nat is symmetric, only the server that received the registrar can send back calls to the phone. Communication between users on two servers is as simple as using t_relay_to(proto:serverip:**port) after you do lookup(location) and no record is found. Suppose I have three or more kamailio servers. If a call comes in and lookup() returns that no record was found, how do I know which server to forward to? Is that a case in which I should replicate the database? You have to add an extra check for the case the call was coming from the other server, not to forward back to it in case of no found again. Thanks for the tip, I hadn't thought of that. Also, you should skip user authentication for calls from the other server (not do authenticate twice). Another aspect to take care is chaining rtpproxy, you have to use the flat for trusting the other server (r, iirc). You can also replicate the registration, but again, it can add troubles to the nat. Look at t_replicate() (in tm module). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013 - more details about Kamailio trainings at http://www.asipto.com - __**_ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users www.sipcentric.com Follow us on twitter @sipcentric http://twitter.com/sipcentric Sipcentric Ltd. Company registered in England Wales no. 7365592. Registered office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham B7 4EJ. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr
Re: [SR-Users] Calling users on another kamailio server
I thought append_hf() didn't take affect (unless you use msg_apply_changes()) until the message left Kamailio too? If that is the case, and msg_apply_changes() is called, doesn't that mean the Path: header from add_path() would be added in that scenario? On 16 September 2013 14:49, Charles Chance charles.cha...@sipcentric.comwrote: It is possible if the edge proxy and registrar are separate, as you say. But if the registrar is at the edge with no separate proxy, add_path() does nothing (because the message never leaves Kamailio for the header to be added). On 16 September 2013 14:42, Peter Dunkley peter.dunk...@crocodilertc.netwrote: Is there any reason it isn't possible? On 16 September 2013 14:21, Charles Chance charles.cha...@sipcentric.com wrote: Hi, Yes, you are right - and I agree, it would be better if this was possible :) Charles On 16 September 2013 14:15, Peter Dunkley peter.dunk...@crocodilertc.net wrote: Hello, It'd be better if the add_path() function could be used here. That way, if using outbound (RFC5626), the flow-token (the userinfo part of the Path-URI) would be present and there would be no need to add the ;received parameter. This would address the one issue remaining for SIP outbound on Kamailio, which is its use without an edge proxy that is separate from the registrar. Regards, Peter On 16 September 2013 14:05, Charles Chance charles.cha...@sipcentric.com wrote: Hi, This sounds like a case for sharing same database, and adding Path before saving incoming register. That way, no need to replicate register message to other servers and all subscribers use the same domain. Add path something like this before calling save(): append_hf(Path: sip:$Ri:$Rp; received=sip:$si:$sp;lr\r\n); msg_apply_changes(); Whichever server receives the incoming invite, will perform lookup and automatically route to the server which received the register. On the proxying server set $du according to received param of route header, add record-route, and then t_relay(). As Daniel said, no need to re-authenticate or perform lookup again. Regards, Charles On Mon, Sep 16, 2013 at 7:34 AM, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 9/12/13 10:08 PM, Brian Wallen wrote: I currently have two independent kamailio servers. I'd like to set them up in a way that user1 on server1 can make a call to user2 on server2. After searching I've come up with two ways that this might be able to be done. Can someone please sanity check these or point me in the right direction? 1. Have one registrar server and convert the other server to a proxy 2. Keep them both as registrars and somehow make them each aware of the users on the other server I like 2 better because if one server went down users on the other server would still be up. The only thing is I don't know how to set the servers up to communicate with each other. the nat can create problems when a server is down - if the nat is symmetric, only the server that received the registrar can send back calls to the phone. Communication between users on two servers is as simple as using t_relay_to(proto:serverip:**port) after you do lookup(location) and no record is found. Suppose I have three or more kamailio servers. If a call comes in and lookup() returns that no record was found, how do I know which server to forward to? Is that a case in which I should replicate the database? You have to add an extra check for the case the call was coming from the other server, not to forward back to it in case of no found again. Thanks for the tip, I hadn't thought of that. Also, you should skip user authentication for calls from the other server (not do authenticate twice). Another aspect to take care is chaining rtpproxy, you have to use the flat for trusting the other server (r, iirc). You can also replicate the registration, but again, it can add troubles to the nat. Look at t_replicate() (in tm module). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013 - more details about Kamailio trainings at http://www.asipto.com - __**_ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users www.sipcentric.com Follow us on twitter @sipcentric http://twitter.com
Re: [SR-Users] Kamailio Websocket and sipml5
Sorry, hit send early by mistake... Kamailio is a SIP proxy and does not do anything with media. You can find lots of comments and discussion about this on this list and sr-dev. Basically, WebRTC mandates a media profile (RTP/SAVPF) that is not supported by most existing clients and servers. As such you need something more than just Kamailio (which is SIP signalling only) to handle this media profile. Regards, Peter On 2 September 2013 03:28, Peter Dunkley peter.dunk...@crocodilertc.netwrote: Hello, On 1 September 2013 14:03, Jason Sia jsi...@gmail.com wrote: I configured kamailio websocket server the same as the configuration in the examples websocket.conf in the git repository. I can successfully register from sipml5 client however when I place a call it shows unsupported media type any ideas? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] unregister user when kamailio looses TCP connection.
I started to implement this. It is in a branch somewhere, but I couldn't get it working. If someone who knows the TCP code better could take a look... Regards, Peter On 26 August 2013 23:28, Olle E. Johansson o...@edvina.net wrote: 27 aug 2013 kl. 08:27 skrev Olle E. Johansson o...@edvina.net: 27 aug 2013 kl. 00:29 skrev Vitaliy Aleksandrov vitalik.v...@gmail.com : Hello, I've made a patch to kamailio-4.0.3 which removes stale registration when kamailio looses an incoming tcp connection. Of course this patch needs more work. Since the are no direct references between user location contacts and tcp connections callback function uses linear search through the whole location table using received field as a key. Can anybody more experienced in kamailio internals check if I chose the right place to get information about lost tcp connections ? Another thing I wanted to ask is maybe somebody can suggest a better way to tie a tcp connection to the user location information without complicating usrloc module by any heavy data structures. If anybody else except me need this It would be great to fix known problems and add it to kamailio. remove_tcp_contacts.patch___ This is something required for outbound too. We need to remove the registration and thus the flow if a connection dies. The problem is that we can manage the connection on another server (edge proxy) too. ...which is why I earlier proposed an event-route for this use-case. /O ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Error message installing kamailio websocket
Hello, What version of libunistring does Ubuntu 10.04 come with? How far away is that from the latest version of libunistring (or even just the versions that are available on other platforms like CentOS, Fedora, or Ubuntu 13.04)? It could just be that you need a later version of libunistring than is available in that three year old version of Ubuntu. Regards, Peter On 23 August 2013 14:32, Kethzer Docteur kethzer...@gmail.com wrote: I got this error message while installing kamailio with websocket CC (gcc) [M websocket.so] ws_frame.o In file included from ws_frame.c:25: /usr/include/unistr.h:189: error: expected ‘;’, ‘,’ or ‘)’ before ‘_UNUSED_PARAMETER_’ /usr/include/unistr.h:259: error: expected ‘;’, ‘,’ or ‘)’ before ‘_UNUSED_PARAMETER_’ make[1]: *** [ws_frame.o] Error 1 make: *** [modules] Error 1 DISTRIB_ID=Ubuntu DISTRIB_RELEASE=10.04 DISTRIB_CODENAME=lucid DISTRIB_DESCRIPTION=Ubuntu 10.04.2 LTS Any Help please On Fri, Aug 23, 2013 at 9:31 AM, Kethzer Docteur kethzer...@gmail.comwrote: DISTRIB_ID=Ubuntu DISTRIB_RELEASE=10.04 DISTRIB_CODENAME=lucid DISTRIB_DESCRIPTION=Ubuntu 10.04.2 LTS On Fri, Aug 23, 2013 at 9:29 AM, Kethzer Docteur kethzer...@gmail.comwrote: CC (gcc) [M websocket.so] ws_frame.o In file included from ws_frame.c:25: /usr/include/unistr.h:189: error: expected ‘;’, ‘,’ or ‘)’ before ‘_UNUSED_PARAMETER_’ /usr/include/unistr.h:259: error: expected ‘;’, ‘,’ or ‘)’ before ‘_UNUSED_PARAMETER_’ make[1]: *** [ws_frame.o] Error 1 make: *** [modules] Error 1 -- Kethzer Docteur -- Kethzer Docteur -- Kethzer Docteur ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] mediaproxy-ng one-way audio (but not all the time)
Hello, I am testing mediaproxy-ng (running on CentOS 6 on Amazon EC2) for WebRTC to non-WebRTC calls and I am getting one-way audio most (but not all) of the time. I always get audio in the WebRTC to non-WebRTC direction. Has anybody had any experience of anything this? I have checked the obvious (all the ports open in iptables and the Amazon security groups). I know that the right (advertised) address is going into the SDP. I am getting an error message out of mediaproxy-ng a lot. But it is there whether the audio works or not. The error I am seeing is: Error generating SRTP session keys. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamailio STUN Server
If you are building from Git master STUN is now a module and is not compiled with STUN=1. Whether you are using Git master or not, the Kamailio STUN implementation is limited. It is a partial implementation designed to run on the same port as SIP for the purposes of signalling NAT traversal. It works best in combination with SIP outbound (RFC 5626). If you are looking to use STUN for media NAT traversal (either in an ICE or non-ICE environment) you probably want to use a different (and more complete) STUN implementation. Regards, Peter On 16 August 2013 10:44, Premchandiran premchandiran.marimu...@plintron.com wrote: Hi All, I am trying kamailio with STUN , I was able to enable STUN using make cfg STUN=1 my question is does kamailio itself STUN server? Since stun request (bind request) is sent from client kamailio is not responding. Somewhere read that kamailio uses liseten port as STUN also. ** ** Thanks and Regards, *Prem Chandiran M*** ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] websocket double record_route()
Are you using outbound? If you are I suggest trying git master rather than the 4.0 branch. Regards, Peter On 31 July 2013 10:53, Vitaliy Aleksandrov vitalik.v...@gmail.com wrote: Hello. I'm trying to configure kamailio as a gateway between Websocket and TCP/TLS transports. When I call record_route() for an initial INVITE that comes via WS and will be forwarded via TCP to a registered UA kamailio inserts only a one record-route header with its IP and transport=ws instead of two record-route headers with both incoming/outgoing transports. This behaviour breaks in-dialog requests routing. rr module parameters are: modparam(rr, enable_full_lr, 1) modparam(rr, append_fromtag, 0) modparam(rr, enable_double_rr, 1) I use: - kamailio-4.0.2 - sipml5 as sip client Can anybody point me in the right direction to understand why it happens ? __**_ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] websocket double record_route()
Have you tried the latest code on the 4.0 branch as I am sure there are fixes that have been made since the last release? Also, if using WebSockets I do strongly recommend you try Git master if you have any problems. There may well be fixes there that haven't been back-ported to the 4.0 branch yet. Regards, Peter On 31 July 2013 11:56, Vitaliy Aleksandrov vitalik.v...@gmail.com wrote: Hello, Thanks for a quick reply. No, outbound module is not loaded. As I understood from RR docs double_rr won't work if outbound module is present. I've added mhomed=1 to my config, but record_route() still inserts only one rr with transport=ws. Are you using outbound? If you are I suggest trying git master rather than the 4.0 branch. Regards, Peter On 31 July 2013 10:53, Vitaliy Aleksandrov vitalik.v...@gmail.comwrote: Hello. I'm trying to configure kamailio as a gateway between Websocket and TCP/TLS transports. When I call record_route() for an initial INVITE that comes via WS and will be forwarded via TCP to a registered UA kamailio inserts only a one record-route header with its IP and transport=ws instead of two record-route headers with both incoming/outgoing transports. This behaviour breaks in-dialog requests routing. rr module parameters are: modparam(rr, enable_full_lr, 1) modparam(rr, append_fromtag, 0) modparam(rr, enable_double_rr, 1) I use: - kamailio-4.0.2 - sipml5 as sip client Can anybody point me in the right direction to understand why it happens ? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] MSRP communication through Kamailio
Hello, If your client supports RFC 4975 for peer-to-peer MSRP then you will see those attributes in the SDP. That doesn't mean that the client supports RFC 4976 MSRP relay. Kamailio is an MSRP relay, not a B2BUA. If you want MSRP B2BUA in Kamailio you'd need to write a whole new module just for that. However, Kamailio isn't really the right architecture for a B2BUA. If your client does not support RFC 4976 you aren't going to be able to use it with Kamailio. In this situation you will need some form of B2BUA (probably an SBC) with MSRP support in it - and that probably means a commercial product with a significant price attached. Regards, Peter On 29 July 2013 12:45, Rajkumar Kanniappan rajkumar.kanniap...@sasken.comwrote: Thanks Peter. So you mean to say kamailio doesnt support B2BUA for msrp communication? My client supports all the required MSRP SDP parameters and configurations are fine. Because I am able to see proper msrp headers in the initial INVITE. Thanks From: sr-users-boun...@lists.sip-router.org [ sr-users-boun...@lists.sip-router.org] On Behalf Of Peter Dunkley [ peter.dunk...@crocodilertc.net] Sent: Monday, July 29, 2013 16:34 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] MSRP communication through Kamailio There are two standards based options for MSRP. Peer-to-peer between clients (RFC 4975) with no servers involved and traffic between clients relayed through a server (RFC 4976). RFC 4976 is used for NAT traversal and policy (security/logging/etc) enforcement. The relay option requires the MSRP client to support additional MSRP request types, procedures, and SDP elements - essentially use of a relay requires the client to support this and be configured for it. There is a third option which is not standards based, that is B2BUA'ing MSRP where a server pretends to be a pair of clients back-to-back. This is what a lot of SBCs do. As a server that is not a B2BUA Kamailio supports MSRP relay as per RFC 4976. However, your MSRP client needs to support RFC 4976 (and be correctly configured) to make use of the relay. Regards, Peter On 29 July 2013 11:20, Rajkumar Kanniappan rajkumar.kanniap...@sasken.com mailto:rajkumar.kanniap...@sasken.com wrote: Hi, Is it possible to make MSRP to pass through kamailio instead of peer to peer? For me, my sip client registers with kamailio without any problem and able to chat with other sip client as well. But the chat communication using MSRP is always peer to peer, because of the SDP negotiation. But I need make the msrp messages to pass through the kamailio server. Please help me in configuring the kamailio. Thanks SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited (Sasken) unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.orgmailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] automatic call termination when websocket connection is closed
Hello, I think the only way to do this would be to use the dialog module. However, as far as I know the appropriate dialog function (dlg_bye) needs to be run from a route where the SIP message is within the dialog in question. In other words, there is no way you could run this from event_route[websocket:closed] because even if you were able to to store the information required to identify the (potentially multiple dialogs) relating to the WebSocket connection (perhaps in a hash-table) you can't actually do anything with these dialogs from there. If this really is a requirement I suspect you're going to need to make some enhancements to the dialog module. Alternatively, you can take advantage of the GRUU and outbound support in Kamailio git master which should allow calls to survive WebSocket connection close and re-establishment. Regards, Peter On 13 June 2013 10:58, Iwan Budi Kusnanto i...@labhijau.net wrote: Related to 'Doing automatic unregister when a WEBSOCKET connection is closed' thread. http://sip-router.1086192.n5.nabble.com/Doing-automatic-unregister-when-a-WEBSOCKET-connection-is-closed-td118083.html Any hint to implement automatic call termination when websocket connection is closed? Some of my ideas: 1. send BYE message from UAC module. Looks like it is an unnecessary complex dirty solution. 2. Utilizing dialog module, but i'm not sure if dialog module can be used to implement this. Any better idea? Thanks. -- Iwan Budi Kusnanto ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Need help in setting up Presence
X-Lite does not support server presence (or didn't last time I looked) - at
least not for 'social' presence, MWI may work.
It works by having the clients exchange SUBSCRIBEs and NOTIFYs with each other
and not doing any PUBLISHes. As such, server presence isn't going to work.
Regards,
Peter
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
On 10 Jun 2013, at 11:05, SamyGo govoi...@gmail.com wrote:
Hi List,
I've been trying to make the Presence thing work with kamailio but the very
basic presence doesn't seem to work. I've tried multiple modules and
different how-tos for running successful presence aware configuration but
seems something is missing. The ultimate goal is to give user
online/busy/offline status updaes to Subscribers.
My Kamailio version is: kamailio 4.0.1 (i386/linux) 55f7de
I've loaded the following modules.
loadmodule presence.so
loadmodule presence_xml.so
loadmodule presence_dialoginfo.so
loadmodule presence_reginfo
loadmodule pua.so
#loadmodule sca.so
loadmodule pua_dialoginfo.so
loadmodule pua_usrloc.so
loadmodule pua_reginfo
The Presence route contains this:
route[PRESENCE] {
if(!is_method(PUBLISH|SUBSCRIBE))
return;
xlog(L_INFO, [$fU@$si:$sp]{$rm} In Presence Route \n);
#!ifdef WITH_PRESENCE
if (!t_newtran())
{
sl_reply_error();
exit;
};
if(is_method(PUBLISH))
{
handle_publish();
t_release();
}
else
if( is_method(SUBSCRIBE))
{
handle_subscribe();
t_release();
}
exit;
#!endif
# if presence enabled, this part will not be executed
if (is_method(PUBLISH) || $rU==$null)
{
sl_send_reply(404, Not here);
exit;
}
return;
}
Then I've my x-lite phone (which is known to work with Presence in Asterisk)
tries to register and subscribe to its own extensions. Nothing happens. The
trace from sipgrep is attached.
Please help me in making presence work.
Thanks,
Sammy
presence_trace.txt
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] is using Gerrit for Kamailio a good idea?
It might be worth taking a look at the Atlassian tool set. These can be deployed locally or used as a hosted application. I believe there is a free option for open-source projects and the hosted option would make management of them much simpler. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd On 10 Jun 2013, at 08:27, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, I don't have any experience with it, maybe others can comment on its usability. From what I could see, it is written on Java, does it have an embedded http server or we need to install a java application server by ourselves? Since we are here, on admin stuff, anyone knowing an app integrating at least: - git web viewer - code review - bug tracker It might be easier to maintain a single system for all of them rather three different ones. Cheers, Daniel On 6/7/13 11:48 AM, Victor Seva wrote: Hello fellow Kamailio users and developers, I would like to throw the idea of use Gerrit[0] for code contributions reviews. This is not a replacement for the bugtracker, Gerrit will help developers at the task of review and merge contributors patches. I'm going to use mainly [1] to try to explain what Gerrit is: Gerrit is a web-based tool that is used for code review. Its main features are the side-by-side difference viewing and inline commenting which makes code reviews quick and simple task. It is used together with Git version control system. Gerrit allows authorized contributors to submit changes to Git repository, after reviews are done. Contributors can get their code reviewed with a little effort, and get their changes quickly through the system. It has a nice interface to review the different patch versions of the change, you can comment any line of code easily and the change has to be approved to be merged. I would like to know your opinion on this subject and your experience using it if you have it. Cheers, Victor [0] http://gerrit-documentation.googlecode.com/svn/Documentation/2.6/intro-quick.html [1] http://qt-project.org/wiki/Gerrit-Introduction ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamailio 4.0.1 websocket configuration
I am not sure if libunistring is available in CentOS 5. It could well be something that was only added in CentOS 6. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd On 10 Jun 2013, at 08:36, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, have you installed libunistring package and its -dev headers? I don't know how is named on centos exactly, the name I gave is for debian/ubuntu. Cheers, Daniel On 5/28/13 12:46 PM, Rupayan Dutta wrote: Hi All, I have build Kamailio 4.0.1 from source in CentOS 5.8(i386 architecture).I followed all instructions from http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory is not generated).I then edit kamailio.config file for websocket support as described in webocket.cfg file in exmples directory.But while starting kamailio it gives following error ERROR: load_module: could not open module : libunistring.so.0: cannot open shared object file: No such file or directory 0(30270) : [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to load module I checked websocket.so file in the specified directory and it is already there.Can you please help me what's wrong with it? please help. Rupayan Dutta ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Why does Kamailio return many Contacts in Contact header in REGISTER response ?
All of them if you just use lookup() and it serial/parallel forks based on q-value if you use t_load_contacts() and t_next_contacts() correctly. This is the correct SIP behaviour as per RFC 3261. Regards, Peter On 30/05/13 11:05, Khoa Pham wrote: @Olle: So when I have incoming calls, which registration does Kamailio respond to ? (What address does it choose to send me INVITE) ? On Thu, May 30, 2013 at 4:44 PM, Jesús Pérez Rubio jesus.pe...@quobis.com mailto:jesus.pe...@quobis.com wrote: You can change the expires time in your kamailio.cfg file: http://kamailio.org/docs/modules/0.9.x/registrar.html#AEN62 2013/5/30 Olle E. Johansson o...@edvina.net mailto:o...@edvina.net 30 maj 2013 kl. 11:38 skrev Khoa Pham onmyway...@gmail.com mailto:onmyway...@gmail.com: When I make REGISTER request to server Kamailio. Kamailio sometimes return me REGISTER 200 OK response with many Contacts in Contact header field Contact: sip:user1@1.1.1.1:58492;transport=TLS;ob;expires=29;received=sip:1.1.1.1:58492;transport=TLS, sip:user1@3.3.3.3:58520;transport=TLS;ob;expires=244, sip:user1@1.1.1.1:58529;transport=TLS;ob;expires=284;received=sip:1.1.1.1:58529;transport=TLS, sip:user1@3.3.3.3:58548;transport=TLS;ob;expires=329, sip:user1@3.3.3.3:58562;transport=TLS;ob;expires=393, sip:user1@1.1.1.1:58571;transport=TLS;ob;expires=483;received=sip:1.1.1.1:58571;transport=TLS, sip:user1@2.2.2.2:58588;transport=TLS;ob;expires=538, sip:user1@1.1.1.1:58600;transport=TLS;ob;expires=587;received=sip:1.1.1.1:58600;transport=TLS, sip:user1@2.2.2.2:58611;transport=TLS;ob;expires=630, sip:user1@1.1.1.1:58624;transport=TLS;ob;expires=670;received=sip:1.1.1.1:58624;transport=TLS, sip:user1@2.2.2.2:58632;transport=TLS;ob;expires=706, sip:user1@1.1.1.1:58650;transport=TLS;ob;expires=826;received=sip:1.1.1.1:58650;transport=TLS, sip:user1@2.2.2.2:58661;transport=TLS;ob;expires=900;+sip.instance=urn:uuid:----7dtrf0a4c;reg-id=1 Why is that ? The SIP standard says that the response should include ALL current registrations. These are registrations that has not expired yet. /O ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com http://www.quobis.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Khoa Pham HCMC University of Science www.fantageek.com http://www.fantageek.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Why does Kamailio return many Contacts in Contact header in REGISTER response ?
That parameter will prevent new registrations when the maximum contacts have been exceeded. It won't get rid of any stale ones you have. Regards, Peter On 30/05/13 13:08, Vitaliy Aleksandrov wrote: Kamailio can do this but it doesn't by default. http://kamailio.org/docs/modules/4.0.x/modules/registrar.html#idp110136 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] [OT] Crocodile Network WebRTC Launch broadcast
Hello, Crocodile is having a launch event this evening for our new SIP network and SDK (making use of Kamailio with SIP over WebSocket and outbound). The launch will be broadcast live from 18:30 BST. Anyone interested can view the broadcast at http://lnx.so/croc (the link currently points to our blog but will go live properly shortly before the broadcast). Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio for Debian blog?
/dev/null || :
%post
/sbin/chkconfig --add turnserver
%preun
/sbin/service turnserver stop /dev/null 21
/sbin/chkconfig --del turnserver
%files
%defattr(-,root,root)
%{_bindir}/turnserver
%{_bindir}/turnadmin
%{_mandir}/man1/turnserver.1.gz
%{_mandir}/man1/turnadmin.1.gz
%dir %attr(-,turnserver,turnserver) %{_sysconfdir}/%{name}
%config(noreplace) %{_sysconfdir}/%{name}/turnserver.conf
%config(noreplace) %{_sysconfdir}/%{name}/turnuserdb.conf
%config(noreplace) %{_sysconfdir}/sysconfig/turnserver
%config %{_sysconfdir}/rc.d/init.d/turnserver
%dir %{_docdir}/%{name}
%{_docdir}/%{name}/INSTALL
%{_docdir}/%{name}/postinstall.txt
%{_docdir}/%{name}/README.turnadmin
%{_docdir}/%{name}/README.turnserver
%{_docdir}/%{name}/schema.sql
%files utils
%defattr(-,root,root)
%{_bindir}/turnutils_peer
%{_bindir}/turnutils_rfc5769check
%{_bindir}/turnutils_stunclient
%{_bindir}/turnutils_uclient
%{_mandir}/man1/turnutils.1.gz
%{_mandir}/man1/turnutils_peer.1.gz
%{_mandir}/man1/turnutils_rfc5769check.1.gz
%{_mandir}/man1/turnutils_stunclient.1.gz
%{_mandir}/man1/turnutils_uclient.1.gz
%{_libdir}/libturnclient.a
%dir %{_includedir}/turn
%{_includedir}/turn/ns_turn_defs.h
%dir %{_includedir}/turn/client
%{_includedir}/turn/client/ns_turn_ioaddr.h
%{_includedir}/turn/client/ns_turn_msg_addr.h
%{_includedir}/turn/client/ns_turn_msg_defs.h
%{_includedir}/turn/client/ns_turn_msg.h
%{_includedir}/turn/client/TurnMsgLib.h
%{_docdir}/%{name}/README.turnutils
%files doc
%{_docdir}/%{name}/TURNServerRESTAPI.pdf
%{_docdir}/%{name}/TurnNetworks.pdf
%dir %{_docdir}/%{name}/html
%{_docdir}/%{name}/html/*
%dir %{_datadir}/%{name}
%{_datadir}/%{name}/*
%changelog
* Fri May 3 2013 Peter Dunkley pe...@dunkley.me.uk
- First version
diff -uprN turnserver-1.8.3.6.orig/centos/turnserver.init turnserver-1.8.3.6/centos/turnserver.init
--- turnserver-1.8.3.6.orig/centos/turnserver.init 1970-01-01 01:00:00.0 +0100
+++ turnserver-1.8.3.6/centos/turnserver.init 2013-05-04 22:54:48.044001115 +0100
@@ -0,0 +1,78 @@
+#!/bin/bash
+#
+# Startup script for TURN Server
+#
+# chkconfig: 345 85 15
+# description: RFC 5766 TURN Server
+#
+# processname: turnserver
+# pidfile: /var/run/turnserver.pid
+# config: /etc/turnserver/turnserver.conf
+#
+### BEGIN INIT INFO
+# Provides: turnserver
+# Required-Start: $local_fs $network
+# Short-Description: RFC 5766 TURN Server
+# Description: RFC 5766 TURN Server
+### END INIT INFO
+
+# Source function library.
+. /etc/rc.d/init.d/functions
+
+turn=/usr/bin/turnserver
+prog=turnserver
+pidfile=/var/run/$prog.pid
+lockfile=/var/lock/subsys/$prog
+user=turnserver
+RETVAL=0
+
+[ -f /etc/sysconfig/$prog ] . /etc/sysconfig/$prog
+
+start() {
+ echo -n $Starting $prog:
+ # there is something at end of this output which is needed to
+ # report proper [ OK ] status in CentOS scripts
+ daemon --pidfile=$pidfile --user=$user $turn $OPTIONS
+ RETVAL=$?
+ echo
+ [ $RETVAL = 0 ] touch $lockfile
+}
+
+stop() {
+ echo -n $Stopping $prog:
+ killproc $turn
+ RETVAL=$?
+ echo
+ [ $RETVAL = 0 ] rm -f $lockfile $pidfile
+}
+
+[ -z $OPTIONS ] OPTIONS=-c /etc/turnserver/turnserver.conf -o --no-stdout-log
+
+# See how we were called.
+case $1 in
+ start)
+ start
+ ;;
+ stop)
+ stop
+ ;;
+ status)
+ status $turn
+ RETVAL=$?
+ ;;
+ restart)
+ stop
+ start
+ ;;
+ condrestart)
+ if [ -f /var/run/$prog.pid ] ; then
+ stop
+ start
+ fi
+ ;;
+ *)
+ echo $Usage: $prog {start|stop|restart|condrestart|status|help}
+ exit 1
+esac
+
+exit $RETVAL
diff -uprN turnserver-1.8.3.6.orig/centos/turnserver.sysconfig turnserver-1.8.3.6/centos/turnserver.sysconfig
--- turnserver-1.8.3.6.orig/centos/turnserver.sysconfig 1970-01-01 01:00:00.0 +0100
+++ turnserver-1.8.3.6/centos/turnserver.sysconfig 2013-05-04 22:52:41.81834 +0100
@@ -0,0 +1,5 @@
+#
+# TURN Server startup options
+#
+
+OPTIONS=-c /etc/turnserver/turnserver.conf -o --no-stdout-log
diff -uprN turnserver-1.8.3.6.orig/configure turnserver-1.8.3.6/configure
--- turnserver-1.8.3.6.orig/configure 2013-05-01 06:49:50.0 +0100
+++ turnserver-1.8.3.6/configure 2013-05-04 22:34:01.322999200 +0100
@@ -183,6 +183,10 @@ if [ -z ${MANPREFIX} ] ; then
MANPREFIX=${PREFIX}
fi
+if [ -z ${CONFPREFIX} ] ; then
+ CONFPREFIX=${PREFIX}/etc
+fi
+
if [ -z ${EXAMPLESDIR} ] ; then
EXAMPLESDIR=${PREFIX}/share/examples/${PORTNAME}
fi
@@ -642,6 +646,7 @@ echo # Makefile
echo PORTNAME = ${PORTNAME} Makefile
echo PREFIX = ${PREFIX} Makefile
echo MANPREFIX = ${MANPREFIX} Makefile
+echo CONFPREFIX = ${CONFPREFIX} Makefile
echo EXAMPLESDIR = ${EXAMPLESDIR} Makefile
echo DOCSDIR = ${DOCSDIR} Makefile
echo LIBDIR = ${LIBDIR} Makefile
diff -uprN turnserver-1.8.3.6.orig/Makefile.in turnserver-1.8.3.6/Makefile.in
--- turnserver-1.8.3.6.orig/Makefile.in 2013-05-01 06:49:50.0 +0100
+++ turnserver-1.8.3.6/Makefile.in 2013-05-04 22:34:08.076994550 +0100
@@ -105,64 +105,63 @@ distclean: clean
### Install all:
install: all ${MAKE_DEPS}
- ${MKDIR
Re: [SR-Users] Doing automatic unregister when a WEBSOCKET connection is closed.
The usrloc function that locates a record by ruid takes in an aorhash value as an argument (and to get the aorhash value you need the AoR). I suppose an alternative function that just needs ruid could be written - but it would be much less efficient as it would have to linearly search all records (unless an additional hash on ruid is added - and a DB index on it too). Regards, Peter I suppose you could write a new function that just Peter Dunkley writes: 3) Use a hash-table to store $ruid, $tU, and $td indexed on $si:$sp 4) Then when event_route[websocket:closed] is called you can retrieve the information from the hash table and call unregister(). Use the $tU and $td you have cached to construct the unregister() URI parameter. why does unregister need $tU and $td params, because $ruid alone uniquely identifies the contact to be unregistered? -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Doing automatic unregister when a WEBSOCKET connection is closed.
When you call unregister() without the new ruid parameter it parses the current SIP message to get information needed to do the unregister(). There is no real SIP message associated with an event_route[] so unregister() will not work. The way to get this working is: 1) Use Git master so that unregister() with ruid is supported 2) Modify save() so that upon successful creation of new contact it copies the ruid created by usrloc into the sip_msg.ruid variable for the current SIP message. This means that on return from save() the $ruid PV will work. 3) Use a hash-table to store $ruid, $tU, and $td indexed on $si:$sp 4) Then when event_route[websocket:closed] is called you can retrieve the information from the hash table and call unregister(). Use the $tU and $td you have cached to construct the unregister() URI parameter. Regards, Peter On Mon, May 6, 2013 at 4:19 PM, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: and by the way, I found another problem when implementing the first method: when calling unregister(location,websocket=$si:$sp) from the event_route[websocket:closed] i get the following error: *[parser/parse_to.c:879] : failed to parse To uri* why it happens and how can i fix it ? (i think it is related to the fact we call unregister from the event_route, but i'm not sure) You can't call unregister from that event_route. Why we can't call unregister from event_route[websocket:closed]? Is it because it is just not implemented yet or we need big changes to implement it? Thanks, -- Iwan Budi Kusnanto ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Doing automatic unregister when a WEBSOCKET connection is closed.
When stating stable and bug-free - my intention was relativly to that feature in kamailio because it seems that it is still under development. Moreover - certainly if i will find any problem while testing i would submit a bug-report and bug-fixes if possible. and by the way, I found another problem when implementing the first method: when calling unregister(location,websocket=$si:$sp) from the event_route[websocket:closed] i get the following error: *[parser/parse_to.c:879] : failed to parse To uri* why it happens and how can i fix it ? (i think it is related to the fact we call unregister from the event_route, but i'm not sure) You can't call unregister from that event_route. 2013/5/3 Peter Dunkley peter.dunk...@crocodile-rcs.com On 03/05/13 14:48, אורן אברהם wrote: I've tried to implement the first method you've stated. it seems ok but i've found a more fundamental problem: The event_route[websocket:closed] is called only when i teminate the sip stack in my browser, but if i close the browser, without a regular disconnect then the w*ebsocket:closed event is not triggered*. (and this is my main target for the question) *how can i make it trigger ?* The websocket:closed event will be triggered when Kamailio next tries to send a WebSocket ping. Whether Kamailio does this at all (and how frequently it does it) depends entirely on how you've configured the WebSocket module. p.s: about the outbound,path method, *is it a stable feature and bug-free ?* I see the default configuration is still using the nathelper hack instead of it. I am using outbound in my deployments now. It is only available in Kamailio Git master. If stable and bug-free is an absolute requirement that you have then I strongly suggest you consider not using any SIP over WebSocket implementation that exists anywhere. This is brand-new technology and you are on the bleeding edge by using it. I would rate the stability of the Kamailio implementation against anything else you can find and I have no stability issues with it - but I strongly recommend you testing it thoroughly yourself and contributing bug-reports and fixes. Regards, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Doing automatic unregister when a WEBSOCKET connection is closed.
On 03/05/13 14:48, ? wrote: I've tried to implement the first method you've stated. it seems ok but i've found a more fundamental problem: The event_route[websocket:closed] is called only when i teminate the sip stack in my browser, but if i close the browser, without a regular disconnect then the w_ebsocket:closed event is not triggered_. (and this is my main target for the question) *how can i make it trigger ?* The websocket:closed event will be triggered when Kamailio next tries to send a WebSocket ping. Whether Kamailio does this at all (and how frequently it does it) depends entirely on how you've configured the WebSocket module. p.s: about the outbound,path method, *is it a stable feature and bug-free ?* I see the default configuration is still using the nathelper hack instead of it. I am using outbound in my deployments now. It is only available in Kamailio Git master. If stable and bug-free is an absolute requirement that you have then I strongly suggest you consider not using any SIP over WebSocket implementation that exists anywhere. This is brand-new technology and you are on the bleeding edge by using it. I would rate the stability of the Kamailio implementation against anything else you can find and I have no stability issues with it - but I strongly recommend you testing it thoroughly yourself and contributing bug-reports and fixes. Regards, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] service uri not found in rls-services document
Last I looked (around February) there were some things in Kamailio presence that aren't in OpenSIPS. For example, the separate notifier processes in both presence and rls (I believe one may have been ported across now by Saul - but not both). There are also extensions relating to the built-in XCAP server (particularly in presence_xml and rls) that are in Kamailio and not OpenSIPS. I know that there are some specific RLS fixes and features in Kamailio that only work when you use the built-in XCAP server. I also made some enhancements to the way pidf-manipulation is handled in Kamailio that I do not think have been ported to OpenSIPS. Basically, while OpenSIPS presence may be more advanced in some ways I am quite sure it is less developed in some others. Having spent a lot of time over the course of around 18 months doing work in Kamailio presence I would strongly resist any attempts to junk what is currently there because I know that I need a lot of the stuff that is in the Kamailio implementation. I don't currently have the time to develop it further at the moment either. I think the junk them and replace with opensips approach would be a disaster. Regards, Peter looks like kamailio presence and especially rls is lagging behind in development as compared to opensips. it properly handles the escapes that i had trouble with, support external references in presence rules, xcal-diff, etc. what should we do about kamailio presence modules? junk them and replace with opensips ones or implement the missing stuff ourselves? i personally don't have resources for the latter. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] RLS Module - Getting NOTIFY for offline users
Hello, You can do this by putting pidf-manipulation documents into the XCAP server to provide hard presence state. Take a look at the example in http://kamailio.org/docs/modules/stable/modules/xcap_server.html to see the special handling you have to do in XCAP for pidf-manipulation documents to get the presence information they contain PUBLISHed. Note: the way this works is different between Kamailio 3.x and Kamailio 4.0. I recommend you use Kamailio 4.0 if you need hard presence state. Regards, Peter Dear Users, We use Kamailio as our SIP proxy and Presence server, leveraging XCAP and RLS functionality. After issuing SUBSCRIBE rls@domain I would like to get presence information for the contacts in the RLS list not only when they are online, but also if they are offline. Is this possible? Currently we are getting NOTIFY only for those contacts that are online (presence open) Thank you in advance, Attila ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] path uri problem
You can use the force_outbound option in the outbound module to make path and rr add flow-tokens even when the client isn't doing outbound. Regards, Peter On 10/04/13 11:44, Juha Heinanen wrote: Iñaki Baz Castillo writes: Anyhow, why is the received stuff required at all? IMHO it is time for dropping custom/proprietary hacks and use rfc 5626 Outbound instead. Otherwise we must live with hacks in lot of places of the code and modules. unfortunately it will take years before most sip clients implement outbound. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] path uri problem
Single server outbound is on my todo list. I have put the details of what is needed here: http://www.kamailio.org/wiki/devel/completing_outbound Don't know if or when I'll get time to do it though. Regards, Peter On 10/04/13 11:54, Juha Heinanen wrote: Peter Dunkley writes: You can use the force_outbound option in the outbound module to make path and rr add flow-tokens even when the client isn't doing outbound. thank for info. it may not work though, if registrar and edge proxy are combined. in my current test, i have two proxies/registrars each serving serving as edge proxy for the other. when ua registers with p1, p1 can force outbound when it forwards register to p2, but it cannot add those when it processes the register itself. perhaps there could be an option to take path and rr flow tokens from pseudo vars instead of the headers? -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] path uri problem
When using a non-outbound client like Jitsi you can keep-alive by getting it to re-REGISTER, OPTIONS ping, or '\r\n' frequently. IMHO that is far better solution than having the server run timers and generate keep-alives. Regards, Peter On 10/04/13 13:53, Klaus Darilion wrote: Who does keep-alive when outbound is used? If it is the client, then there still must be some tweaks in the server as the non-outbound client will not send keep-alive. regards klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] usrloc + multimaster replication
Hi, The timer to delete records only runs if the timer_interval modparam is 0. This means you can have multiple active registrars off the same database table and use some external process to manage deletion of records (for example, a stored procedure triggered from a cron job). Regards, Peter On 08/04/13 09:52, Alex Balashov wrote: I know I'm revisiting a problem that has been discussed in multiple threads from various angles, so I might be rightly accused of laziness in neglecting to research them all. All the same: I have proxy1 and proxy2 writing to database A and database B, respectively. Database A and database B are active-active masters, synchronised via some replication system attached to the underlying DB technology. The 'location' table is also replicated this way. We know that 'usrloc' doesn't work so well with this: one instance of Kamailio will periodically delete the other's contacts, even if they have a nonlocal SIP domain. Is there any db_mode that can be used (other than 0/purely in-memory) to make this work right? Or is that the essential problem that p_usrloc is written to solve? Thanks, -- Alex ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] usrloc + multimaster replication
As far as I can tell (when db_mode=3) the timer function calls synchronize_all_udomains(), which calls get_act_time() and then db_timer_udomain() in a loop, and db_timer_udomain() just does a database delete. Regards, Peter On 08/04/13 10:27, Alex Balashov wrote: On 04/08/2013 04:58 AM, Peter Dunkley wrote: Hi, The timer to delete records only runs if the timer_interval modparam is 0. This means you can have multiple active registrars off the same database table and use some external process to manage deletion of records (for example, a stored procedure triggered from a cron job). Peter, One question that arises with this solution is: what is the full effect of disabling timer? Is it limited merely to turning off automatic deletion of expired contacts? Or are there other 'synchronisation' tasks that also don't happen as a result? I have trouble imagining what those would be in db_mode 3, but still I wonder. Cheers, -- Alex ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] path uri problem
Thanks for the bug-fix as the parsing should always work, but I am having difficulty understanding the configuration in use here. This particular parsing problem only occurs when outbound is enabled in registrar and you have a received parameter to the Path: header. However, when using outbound I would not expect there to be a received parameter on the Path: header as the outbound flow-token does the same job. Regards, Peter i find and fixed the bug that caused parsing of path uri to fail when save() was called. it still holds true that the path uri generated by add_path_received() is syntactically bogus. if the receiver is not kamailio, parsing of path uri would thus most likely fail. what should be done about it? does kamailio support escaped chars in uri param values? -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Help with SIP over Websocket audio call: 405 Method Not Allowed
Hi,
There is a much simpler WebSocket Kamailio configuration file in the
examples directory in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/welcome.cfg;h=4bde0ae71be6f8da6a5bd3089d5af57569cc0178;hb=HEAD
It doesn't have accounting or any of the other advanced stuff from the
example you provided a link to.
If you can get calls to work with the simple configuration then you know
your problems aren't related to WebSockets, and instead are related to
something else within the configuration.
Regards,
Peter
Peter,
Thank you. By changing the method_filtering modparam to 0 (it was
actually 1), I am now able to make it past this, and the INVITE is
processed over WS transport. However, the audio call is still not
completing.
I am seeing a 180 Ringing message for a while, followed by a 408
Request
Timeout. Nothing is showing ringing on the remote browser with JsSIP
tryit.
The only clues I can see in /var/log/syslog hae to do with Accounting DB.
I am using MySQL.
Note that I can do SIP User Agent client calling just fine between these
two same users, and using the JsSIP Tryit app I can also do 'chat'
messaging. Just can't do audio call. Here is /var/log/syslog:
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: NOTICE: acc
[acc.c:275]: ACC: call missed:
timestamp=1364588167;method=INVITE;from_tag=v71r89q4si;to_tag=mnspl1i563;call_id=8rcjevfvgid74ep1s8rc;code=408;reason=Request
Timeout;src_user=brad;src_domain=xxx.net
;src_ip=172.10.200.149;dst_ouser=joe;dst_user=tmgcpvap;dst_domain=7dq4kria04ks.invalid
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: db_mysql
[km_dbase.c:122]: driver error on query: Unknown column 'src_user' in
'field list'
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: core
[db_query.c:235]: error while submitting query
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: acc
[acc.c:404]:
failed to insert into database
On Thu, Mar 28, 2013 at 9:26 PM, Peter Dunkley
peter.dunk...@crocodile-rcs.com wrote:
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the Allow: header in the
REGISTER.
You can check this by looking at the REGISTER request in a trace and by
inspecting the location records stored in Kamailio (use the ul.dump
command in kamctl for this).
You can disable method filtering in the Kamailio registrar module by
ensuring that the method_filtering modparam is set to 0 (or just not
set
at all as disabled is the default). Doing this should prevent lookup()
ever returning -2.
Regards,
Peter
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket
support,
and
the users can register using the JsSIP Tryit sample WebRTC
application.
They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over
websockets module.
However, after registration, the users can't place an audio call. I
see
no
ringing on the remote browser. Can anyone help with clues or debug?
In
Debug log I can see the websocket ws_frame.c decode the websocket
message
into SIP, and I see normal SIP call flow for an INVITE. However,
nothing
indicating a call.
I ran 'ngrep -p -w -W byline port ' (WS port) and see that I'm
getting
an error response to browser UA of 405: Method Not Allowed. I've
isolated it down to the this snippet in the kamailio.cfg for
route[LOCATION]:
$avp(oexten) = $rU;
if (!lookup(location)) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply(404, Not Found);
exit;
case -2:
send_reply(405, TEST: Method Not
Allowed);
exit;
}
}
The switch case is returning -2, for some reason.
Any help in debugging this appreciated.
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--
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Crocodile RCS Ltd
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Re: [SR-Users] Help with SIP over Websocket audio call: 405 Method Not Allowed
Sorry,
Wrong link. The correct one is:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/websocket.cfg;h=4176af0a86985dc88d768b31f4ebe4021abb093f;hb=HEAD
Peter
Hi,
There is a much simpler WebSocket Kamailio configuration file in the
examples directory in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/welcome.cfg;h=4bde0ae71be6f8da6a5bd3089d5af57569cc0178;hb=HEAD
It doesn't have accounting or any of the other advanced stuff from the
example you provided a link to.
If you can get calls to work with the simple configuration then you know
your problems aren't related to WebSockets, and instead are related to
something else within the configuration.
Regards,
Peter
Peter,
Thank you. By changing the method_filtering modparam to 0 (it was
actually 1), I am now able to make it past this, and the INVITE is
processed over WS transport. However, the audio call is still not
completing.
I am seeing a 180 Ringing message for a while, followed by a 408
Request
Timeout. Nothing is showing ringing on the remote browser with JsSIP
tryit.
The only clues I can see in /var/log/syslog hae to do with Accounting
DB.
I am using MySQL.
Note that I can do SIP User Agent client calling just fine between these
two same users, and using the JsSIP Tryit app I can also do 'chat'
messaging. Just can't do audio call. Here is /var/log/syslog:
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: NOTICE: acc
[acc.c:275]: ACC: call missed:
timestamp=1364588167;method=INVITE;from_tag=v71r89q4si;to_tag=mnspl1i563;call_id=8rcjevfvgid74ep1s8rc;code=408;reason=Request
Timeout;src_user=brad;src_domain=xxx.net
;src_ip=172.10.200.149;dst_ouser=joe;dst_user=tmgcpvap;dst_domain=7dq4kria04ks.invalid
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: db_mysql
[km_dbase.c:122]: driver error on query: Unknown column 'src_user' in
'field list'
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: core
[db_query.c:235]: error while submitting query
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: acc
[acc.c:404]:
failed to insert into database
On Thu, Mar 28, 2013 at 9:26 PM, Peter Dunkley
peter.dunk...@crocodile-rcs.com wrote:
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the Allow: header in the
REGISTER.
You can check this by looking at the REGISTER request in a trace and by
inspecting the location records stored in Kamailio (use the ul.dump
command in kamctl for this).
You can disable method filtering in the Kamailio registrar module by
ensuring that the method_filtering modparam is set to 0 (or just not
set
at all as disabled is the default). Doing this should prevent lookup()
ever returning -2.
Regards,
Peter
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket
support,
and
the users can register using the JsSIP Tryit sample WebRTC
application.
They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip
over
websockets module.
However, after registration, the users can't place an audio call. I
see
no
ringing on the remote browser. Can anyone help with clues or debug?
In
Debug log I can see the websocket ws_frame.c decode the websocket
message
into SIP, and I see normal SIP call flow for an INVITE. However,
nothing
indicating a call.
I ran 'ngrep -p -w -W byline port ' (WS port) and see that I'm
getting
an error response to browser UA of 405: Method Not Allowed. I've
isolated it down to the this snippet in the kamailio.cfg for
route[LOCATION]:
$avp(oexten) = $rU;
if (!lookup(location)) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply(404, Not Found);
exit;
case -2:
send_reply(405, TEST: Method Not
Allowed);
exit;
}
}
The switch case is returning -2, for some reason.
Any help in debugging this appreciated.
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list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
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Re: [SR-Users] Iimitations of xcap server in kamailio
There are a number of core parameter and modparam values you need to increase from their defaults in order to be able to handle large HTTP requests (and therefore large XCAP documents). For example, * tcp_rd_buf_size (core parameter) * tcp_wq_max (core parameter) * sql_buffer_size (core parameter) * buf_size (xcap_server modparam) Regards, Peter Has anyone else noticed issues when adding a large number of contacts to your buddy list? I was using Jitsi and at about 40 contacts I was encountering issues in that the subsequent contacts would not get added to my buddy list. Ttyl, Dave ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Help with SIP over Websocket audio call: 405 Method Not Allowed
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the Allow: header in the
REGISTER.
You can check this by looking at the REGISTER request in a trace and by
inspecting the location records stored in Kamailio (use the ul.dump
command in kamctl for this).
You can disable method filtering in the Kamailio registrar module by
ensuring that the method_filtering modparam is set to 0 (or just not set
at all as disabled is the default). Doing this should prevent lookup()
ever returning -2.
Regards,
Peter
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket support,
and
the users can register using the JsSIP Tryit sample WebRTC application.
They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over
websockets module.
However, after registration, the users can't place an audio call. I see
no
ringing on the remote browser. Can anyone help with clues or debug? In
Debug log I can see the websocket ws_frame.c decode the websocket message
into SIP, and I see normal SIP call flow for an INVITE. However, nothing
indicating a call.
I ran 'ngrep -p -w -W byline port ' (WS port) and see that I'm getting
an error response to browser UA of 405: Method Not Allowed. I've
isolated it down to the this snippet in the kamailio.cfg for
route[LOCATION]:
$avp(oexten) = $rU;
if (!lookup(location)) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply(404, Not Found);
exit;
case -2:
send_reply(405, TEST: Method Not
Allowed);
exit;
}
}
The switch case is returning -2, for some reason.
Any help in debugging this appreciated.
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Peter Dunkley
Technical Director
Crocodile RCS Ltd
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Re: [SR-Users] SIP over WebSockets error: bad request
Hello, Check that you have tcp_accept_no_cl=yes in your configuration. This is required for Kamailio to parse messages over TCP which do not have Content-Length: headers (including HTTP requests). If that doesn't work try running Kamailio with debug output at its highest level (you'll need to output debug to stderr when doing this as syslog won't capture it all) so that more detailed debug can be provided to track this down. Regards, Peter Hi, Wondering if anyone could please help me debug what is wrong with my kamailio.cfg for Websockets support. I'm using JsSIP sample application, and I've compiled Kamailio 4.0 for WS support. My config is borrowed from link this link [1]. Wondering if anyone can help me debug this. Error is bad request, from syslog: Mar 26 19:26:42 ace /usr/local/sbin/kamailio[25774]: ERROR: core [tcp_read.c:1296]: ERROR: tcp_read_req: bad request, state=7, error=4 buf:#012GET / HTTP/1.1#015#012Upgrade: websocket#015#012Connection: Upgrade#015#012Host: sip.XXX.net:#015#012Origin: http://sip.XXX.net#015#012Sec-WebSocket-Protocol: sip#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Sec-WebSocket-Key: p2cM0XbAejvloY1h+pACIw==#015#012Sec-WebSocket-Version: 13#015#012Sec-WebSocket-Extensions: x-webkit-deflate-frame#015#012Cookie: _pk_id.6.4c1d=3ee006d4f8af735c.1363914085.2.1364326003.1363915087.; _pk_ses.6.4c1d=*#015#012#015#012#012parsed:#012GET / HTTP/1.1#015#012Upgrade: websocket#015#012Connection: Upgrade#015#012Host: sip.XXX.net:#015#012Origin: http://sip.XXX.net#015#012Sec-WebSocket-Protocol: sip#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Sec-WebSocket-Key: p2cM0XbAejvloY1h+pACIw==#015#012Sec-WebSocket-Version: 13#015#012Sec-WebSocket-Extensions: x-webkit-deflate-frame#015#012Cookie: _pk_id.6.4c1d=3ee006d4f8af735c.1363914085.2.1364326003.1363915087.; _pk_ses.6.4c1d=*#015#012#015#012 Any ideas? Can send my kamailio.cfg if necessary. [1] Kamailio sample config for WebSockets https://gist.github.com/jesusprubio/4066845 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SIP over WebSockets error: bad request
tcp_accept_no_cl=yes should always be used when Kamailio is receiving HTTP traffic (the event_route[xhttp:request] is used) as HTTP/1.1 does not use the Content-Length: header if there is no message body (as in the case of the WebSocket handshake). Even when there is a body Content-Length: is not needed if HTTP/1.0 backwards compatibility is not required. SIP is different. A SIP message over TCP must always contain a Content-Length: header. When there is no body the Content-Length: is 0. JsSIP is doing exactly the right thing here. It should not be putting Content-Length: into that WebSocket handshake. Regards, Peter Hello Peter, Thank you for the quick reply. Yes, that did the trick. Changing tcp_accept_no_cl to value of no was the resolution. It seems that the JsSIP Tryit sample code, which I was trying to hack, doesn't use Content-Length in the header. I wonder if this should be changed to set the Content-Length in the HTTP header. At any rate, thanks again. I'm now registered, which is what I was looking to do. Brad On Tue, Mar 26, 2013 at 4:08 PM, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: Hello, Check that you have tcp_accept_no_cl=yes in your configuration. This is required for Kamailio to parse messages over TCP which do not have Content-Length: headers (including HTTP requests). If that doesn't work try running Kamailio with debug output at its highest level (you'll need to output debug to stderr when doing this as syslog won't capture it all) so that more detailed debug can be provided to track this down. Regards, Peter Hi, Wondering if anyone could please help me debug what is wrong with my kamailio.cfg for Websockets support. I'm using JsSIP sample application, and I've compiled Kamailio 4.0 for WS support. My config is borrowed from link this link [1]. Wondering if anyone can help me debug this. Error is bad request, from syslog: Mar 26 19:26:42 ace /usr/local/sbin/kamailio[25774]: ERROR: core [tcp_read.c:1296]: ERROR: tcp_read_req: bad request, state=7, error=4 buf:#012GET / HTTP/1.1#015#012Upgrade: websocket#015#012Connection: Upgrade#015#012Host: sip.XXX.net:#015#012Origin: http://sip.XXX.net#015#012Sec-WebSocket-Protocol: sip#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Sec-WebSocket-Key: p2cM0XbAejvloY1h+pACIw==#015#012Sec-WebSocket-Version: 13#015#012Sec-WebSocket-Extensions: x-webkit-deflate-frame#015#012Cookie: _pk_id.6.4c1d=3ee006d4f8af735c.1363914085.2.1364326003.1363915087.; _pk_ses.6.4c1d=*#015#012#015#012#012parsed:#012GET / HTTP/1.1#015#012Upgrade: websocket#015#012Connection: Upgrade#015#012Host: sip.XXX.net:#015#012Origin: http://sip.XXX.net#015#012Sec-WebSocket-Protocol: sip#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Sec-WebSocket-Key: p2cM0XbAejvloY1h+pACIw==#015#012Sec-WebSocket-Version: 13#015#012Sec-WebSocket-Extensions: x-webkit-deflate-frame#015#012Cookie: _pk_id.6.4c1d=3ee006d4f8af735c.1363914085.2.1364326003.1363915087.; _pk_ses.6.4c1d=*#015#012#015#012 Any ideas? Can send my kamailio.cfg if necessary. [1] Kamailio sample config for WebSockets https://gist.github.com/jesusprubio/4066845 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Fwd: kamailio rls mod
Jitsi does not currently support RLS so this will not work at all. The only open-source SIP client I know of with RLS support is Blink (for Mac only at the moment). Blink won't work with Kamailio at the moment either. This is because there is a huge problem with the SIP SIMPLE specifications in that they don't define enough for people to be able to create inter-operable implementations with just the information in the specifications. I plan to make Kamailio work with Blink presence in the future but I need a non-Mac (preferably Linux) implementation that supports this first (and then I need the time to actually do it). Regards, Peter On 06/03/13 07:53, Dmytro Bogovych wrote: AFAIK after 1) Jitsi should send subscribe message to xcap defined list. After this kamailio can send individual subscribe packets in backend. But as you described - Jitsi does not send this magic subscribe. Problem application is Jitsi - not kamailio. I spend couple of weeks diving into xcap/rls stuff and had the same problem with the jitsi. Finally i end up with own softphone code (based on resiprocate) - it works ok. I'd suggest you to try with bria or blink softphones - maybe they will perform better. If anyone got rls working with jitsi - please share your experience... On Tue, Mar 5, 2013 at 4:46 PM, Aleksandrs Semenenko asemene...@ftctele.com mailto:asemene...@ftctele.com wrote: Hello, I'm looking for assistance in RLS module setup for kamailio 3.3.2 The problem is that I need to handle resource-list updates on my kamalio server. For testing purposes I use Jitsi client. Expected: 1) When a new contact is added to Jitsi or an old one is removed, Jitsi sends updated resource-list XML to caps. - This works fine, I can see PUT request in Wireshark and I can see following log from kamailio.cfg: xlog(= xhttp put: refreshing resource-list for $var(uri)\n); rls_update_subs($var(uri), presence); which hopefully means that rls_update_subs function has been called. 2) After this function was called, I expect SUBSCRIBE messages in backend, not in Jitsi, but I still see Jitsi sending SUBSCRIBE for each contact. What am I doing wrong? Any help would be useful! Regards, Alex. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] installing v4.0.x from git
On 06/03/13 11:26, Daniel-Constantin Mierla wrote: On 3/5/13 9:48 AM, Olle E. Johansson wrote: It's already in the README :-) ok, still missing in the wiki, though :) The Debian-specific package name for libunistring shouldn't be in the README. Otherwise we should include the package name for Fedora, CentOS, FreeBSD, etc, and list which OS and version don't have packages available at all (for example libunistring is in CentOS 6 but not CentOS 5). Such a list would end up bigger than the rest of the README, always hopelessly out-of-date, and then need to be repeated for every single module with external dependencies. Regards, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio only allowing traffic with machines in the dispatcher.cfg?
Call ds_is_from_list([groupid]) when you receive a request. If it returns true the request came from one of the members of the group (and you can proceed), if it returns false you can reject it and drop the request. Regards, Peter On 18/02/13 10:58, Benjamin Henrion wrote: Hi, Does someone know/uses a simple rule so that Kamailio only exchanges traffic with machines in the dispatcher? Best, -- Benjamin Henrion bhenrion at ffii.org FFII Brussels - +32-484-566109 - +32-2-3500762 In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] siptrace/sipcapture and MSRP
Hello, Is there any reason why I couldn't use the existing siptrace() function in event_route[msrp:frame-in] to relay all MSRP messages to a SIP Capture node? Will the SIP Capture node handle receiving these messages? What will end up in the WebHomer display in these cases? Regards, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] siptrace/sipcapture and MSRP
Also, what about stuff from event_route[xhttp:request]? On 15/02/13 14:45, Peter Dunkley wrote: Hello, Is there any reason why I couldn't use the existing siptrace() function in event_route[msrp:frame-in] to relay all MSRP messages to a SIP Capture node? Will the SIP Capture node handle receiving these messages? What will end up in the WebHomer display in these cases? Regards, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] siptrace/sipcapture and MSRP
I know that the HEP3 protocol is designed to handle different message types, so I am curious as to whether the implementation of the capture node can handle it. Also, if the capture node can handle it and it does go into the DB will it cause problems for webhomer, or will webhomer simply ignore non-SIP entries (which would be absolutely fine for my use-case)? Regards, Peter On 15/02/13 15:22, Dragos Dinu wrote: Hi, As far as I know, although making siptrace encapsulate different types of messages is not something very difficult, the sipcapture module is developed to write SIP messages into database. Webhomer 3 reads SIP-related data from DB, so it displays only SIP. Regards, Dragos On 02/15/2013 04:45 PM, Peter Dunkley wrote: Hello, Is there any reason why I couldn't use the existing siptrace() function in event_route[msrp:frame-in] to relay all MSRP messages to a SIP Capture node? Will the SIP Capture node handle receiving these messages? What will end up in the WebHomer display in these cases? Regards, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] siptrace/sipcapture and MSRP
OK. So sipcapture parses the messages itself instead of using the Kamailio's message parsing (which does handle HTTP and MSRP)? Regards, Peter On 15/02/13 15:47, Dragos Dinu wrote: Sipcapture can't handle it, because it parses the data as SIP message. If sipcapture would be able to handle it and get it into the DB, Webhomer will be able to show it, because it shows all the entries that you write into the DB. You can, for example, write the MSRP message into the 'msg' column of the table and leave all the other sip-related columns blank. Webhomer will display the data just like that. Think of Webhomer as a user-interface to see everything that is in there (and to search data). Hope it helps, Dragos On 02/15/2013 05:30 PM, Peter Dunkley wrote: I know that the HEP3 protocol is designed to handle different message types, so I am curious as to whether the implementation of the capture node can handle it. Also, if the capture node can handle it and it does go into the DB will it cause problems for webhomer, or will webhomer simply ignore non-SIP entries (which would be absolutely fine for my use-case)? Regards, Peter On 15/02/13 15:22, Dragos Dinu wrote: Hi, As far as I know, although making siptrace encapsulate different types of messages is not something very difficult, the sipcapture module is developed to write SIP messages into database. Webhomer 3 reads SIP-related data from DB, so it displays only SIP. Regards, Dragos On 02/15/2013 04:45 PM, Peter Dunkley wrote: Hello, Is there any reason why I couldn't use the existing siptrace() function in event_route[msrp:frame-in] to relay all MSRP messages to a SIP Capture node? Will the SIP Capture node handle receiving these messages? What will end up in the WebHomer display in these cases? Regards, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] siptrace/sipcapture and MSRP
Actually, the sipcapture module does use the Kamailio message parsing. It looks like only a few small changes will be required to make it recognise and handle HTTP and MSRP. I will look into doing this. Regards, Peter On 15/02/13 15:56, Peter Dunkley wrote: OK. So sipcapture parses the messages itself instead of using the Kamailio's message parsing (which does handle HTTP and MSRP)? Regards, Peter On 15/02/13 15:47, Dragos Dinu wrote: Sipcapture can't handle it, because it parses the data as SIP message. If sipcapture would be able to handle it and get it into the DB, Webhomer will be able to show it, because it shows all the entries that you write into the DB. You can, for example, write the MSRP message into the 'msg' column of the table and leave all the other sip-related columns blank. Webhomer will display the data just like that. Think of Webhomer as a user-interface to see everything that is in there (and to search data). Hope it helps, Dragos On 02/15/2013 05:30 PM, Peter Dunkley wrote: I know that the HEP3 protocol is designed to handle different message types, so I am curious as to whether the implementation of the capture node can handle it. Also, if the capture node can handle it and it does go into the DB will it cause problems for webhomer, or will webhomer simply ignore non-SIP entries (which would be absolutely fine for my use-case)? Regards, Peter On 15/02/13 15:22, Dragos Dinu wrote: Hi, As far as I know, although making siptrace encapsulate different types of messages is not something very difficult, the sipcapture module is developed to write SIP messages into database. Webhomer 3 reads SIP-related data from DB, so it displays only SIP. Regards, Dragos On 02/15/2013 04:45 PM, Peter Dunkley wrote: Hello, Is there any reason why I couldn't use the existing siptrace() function in event_route[msrp:frame-in] to relay all MSRP messages to a SIP Capture node? Will the SIP Capture node handle receiving these messages? What will end up in the WebHomer display in these cases? Regards, Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Web softphone using websockets against Kamailio
Hello, This topic has been covered many times on the Kamailio/SIP Router lists and the answers you require can be found by searching the lists. I will briefly recap: * SIP signalling and media are totally separate things * SIP over WebSockets will allow an HTML5 based client to exchange signalling information with standard soft-phones and hard-phones. However, this does not mean that the media will interwork * HTML5 media streaming uses WebRTC. WebRTC mandates the use of the RTP/SAVPF media profile which is not yet supported by many soft-phones, hard-phones, or media servers This means that, provided you have configured Kamailio and sipml5 correctly, you can get the signalling part of a call working but you will almost certainly have media issues. Kamailio is a SIP signalling device, not a media device, so fixing these media issues is outside of the scope of Kamailio. You do have a few of options with regards to the media but they are limited at the moment. * You can try and find a phone/client that supports RTP/SAVPF (the only ones I know of are the Doubango clients and they sometimes have other issues). * You can use a media server to convert from RTP/SAVPF (Asterisk supports this in theory, but does have issues - I believe there are fixes in the latest Asterisk trunk if you want to compile it yourself - and there may be some non-open-source media servers available). * You can use an RTP Proxy to convert from RTP/SAVPF (erlrtpproxy has this feature on the roadmap, but I don't know whether it is available yet). As for IE support, your guess is as good as mine. Microsoft has its own agenda and has recently been pushing the competing CU-Web-RTC specification. I have a personal opinion about how things will eventually evolve but no facts to share here - I don't believe anyone outside of Microsoft could tell you what will actually happen with IE. Regards, Peter On Mon, 2013-01-28 at 20:45 +0200, Pirjo Ahvenainen wrote: Greetings gurus! I'm playing with an idea to create a web based softphone (html5 + no installations for the end user) and use Kamailio's websocket module for backend. I'd love to hear about your comments, challenges and successes using such configuration. Is it a feasible way to construct a softphone even today when even IE9 does not support websockets, as such? I'm sure IE9 will end up in specs as a must-support platform. A collegue tried using sipml5 with webrtc against a SnomONE pbx (I know... ;)), and said there's no way it can work, but I'm not convinced the idea itself wouldn't work. It would help me lots if I could make a simple example using Kamailio with SIP over websockets, can you comment on how much effort do I need on Kamailio side to make this work? Do I need off-default config scripting, or is it enough to just set up the module and set the parameters? And even with the risk of stepping a little off topic, if anyone has worked on web based softphones, I'd love to hear if you can recommend on how to approach this. Cheers, Pirjo ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websockets WSS problem with NOTIFY
I am not sure how to investigate this. It sounds like it might be a TLS related problem (or a WebSocket/TLS interworking problem in Kamailio). I don't know anything about the Kamailio TLS implementation - I just drop WebSocket frames into it as required. I did do (a little) WSS testing and saw no problems myself. Regards, Peter On 23 Jan 2013, at 22:12, Pete Kelly pke...@gmail.com wrote: Hi, I am having an issue at the moment with SIP NOTIFY messages being sent from Kamailio (latest git master) over wss transport I am getting reports from the receiving end saying Compressed bit must be 0 if no negotiated deflate-frame extension The only reference I can find to it is at the following URL... where the problem was caused by the server miscalculating the size of the msg: http://stackoverflow.com/questions/12308728/compressed-bit-must-be-0-when-sending-a-message-to-websocket-client Does anyone have any suggestions as to how I could debug this within Kamailio? It sounds like Kamailio may be sending some incorrect packet information but I am unsure at this point. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websockets WSS problem with NOTIFY
I shod also add that the Kamailio WebSocket implementation does not support any extensions. So unless the deflate frame extension is implicit for TLS it will not be negotiated. Further, the implementation does not set any compressed bits and all unused flags etc should be zeroed automatically - but I will look at the code later. Peter On 24 Jan 2013, at 09:05, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: I am not sure how to investigate this. It sounds like it might be a TLS related problem (or a WebSocket/TLS interworking problem in Kamailio). I don't know anything about the Kamailio TLS implementation - I just drop WebSocket frames into it as required. I did do (a little) WSS testing and saw no problems myself. Regards, Peter On 23 Jan 2013, at 22:12, Pete Kelly pke...@gmail.com wrote: Hi, I am having an issue at the moment with SIP NOTIFY messages being sent from Kamailio (latest git master) over wss transport I am getting reports from the receiving end saying Compressed bit must be 0 if no negotiated deflate-frame extension The only reference I can find to it is at the following URL... where the problem was caused by the server miscalculating the size of the msg: http://stackoverflow.com/questions/12308728/compressed-bit-must-be-0-when-sending-a-message-to-websocket-client Does anyone have any suggestions as to how I could debug this within Kamailio? It sounds like Kamailio may be sending some incorrect packet information but I am unsure at this point. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websockets WSS problem with NOTIFY
Hi, I've done some checking online and in the code. The compressed bit is defined in draft-ietf-hybi-permessage-compression and uses the RSV1 bit from the WebSocket frame header. As per RFC 6455 the Kamailio WebSocket implementation is careful to leave RSV1, RSV2, and RSV3 with values of 0. As this part of the code is identical for WS and WSS connections can you confirm that it works correctly for WS? Regards, Peter On Thu, 2013-01-24 at 09:09 +, Peter Dunkley wrote: I shod also add that the Kamailio WebSocket implementation does not support any extensions. So unless the deflate frame extension is implicit for TLS it will not be negotiated. Further, the implementation does not set any compressed bits and all unused flags etc should be zeroed automatically - but I will look at the code later. Peter On 24 Jan 2013, at 09:05, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: I am not sure how to investigate this. It sounds like it might be a TLS related problem (or a WebSocket/TLS interworking problem in Kamailio). I don't know anything about the Kamailio TLS implementation - I just drop WebSocket frames into it as required. I did do (a little) WSS testing and saw no problems myself. Regards, Peter On 23 Jan 2013, at 22:12, Pete Kelly pke...@gmail.com wrote: Hi, I am having an issue at the moment with SIP NOTIFY messages being sent from Kamailio (latest git master) over wss transport I am getting reports from the receiving end saying Compressed bit must be 0 if no negotiated deflate-frame extension The only reference I can find to it is at the following URL... where the problem was caused by the server miscalculating the size of the msg: http://stackoverflow.com/questions/12308728/compressed-bit-must-be-0-when-sending-a-message-to-websocket-client Does anyone have any suggestions as to how I could debug this within Kamailio? It sounds like Kamailio may be sending some incorrect packet information but I am unsure at this point. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websockets WSS problem with NOTIFY
Have you checked to see if there are any known bugs in the browser you are using? As the WebSocket message compression stuff is still draft the browser implementation probably won't be complete or fully tested yet. As I said, the Kamailio WebSocket implementation does not support any extensions and all the reserved bits are 0'd. So I don't think it is likely that the compressed bit is set to 1 at all. The only other thing I can suggest is capturing your TLS traffic with WireShark and importing the certificates into it so you can decode the packets. At that point you should be able to look at the binary of the frame and see if the compressed bit is set or not. Regards, Peter On Thu, 2013-01-24 at 13:45 +, Pete Kelly wrote: Hi Peter I can confirm it works correctly for WS and not WSS, and it appears to be only the NOTIFY request in the direction of Kamailio UAC. INVITE requests in the direction of Kamailio UAC are fine. I've tried it with the tls tls_disable_compression flag set to both 0 and 1 Pete On 24 January 2013 09:53, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: Hi, I've done some checking online and in the code. The compressed bit is defined in draft-ietf-hybi-permessage-compression and uses the RSV1 bit from the WebSocket frame header. As per RFC 6455 the Kamailio WebSocket implementation is careful to leave RSV1, RSV2, and RSV3 with values of 0. As this part of the code is identical for WS and WSS connections can you confirm that it works correctly for WS? Regards, Peter On Thu, 2013-01-24 at 09:09 +, Peter Dunkley wrote: I shod also add that the Kamailio WebSocket implementation does not support any extensions. So unless the deflate frame extension is implicit for TLS it will not be negotiated. Further, the implementation does not set any compressed bits and all unused flags etc should be zeroed automatically - but I will look at the code later. Peter On 24 Jan 2013, at 09:05, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: I am not sure how to investigate this. It sounds like it might be a TLS related problem (or a WebSocket/TLS interworking problem in Kamailio). I don't know anything about the Kamailio TLS implementation - I just drop WebSocket frames into it as required. I did do (a little) WSS testing and saw no problems myself. Regards, Peter On 23 Jan 2013, at 22:12, Pete Kelly pke...@gmail.com wrote: Hi, I am having an issue at the moment with SIP NOTIFY messages being sent from Kamailio (latest git master) over wss transport I am getting reports from the receiving end saying Compressed bit must be 0 if no negotiated deflate-frame extension The only reference I can find to it is at the following URL... where the problem was caused by the server miscalculating the size of the msg: http://stackoverflow.com/questions/12308728/compressed-bit-must-be-0-when-sending-a-message-to-websocket-client Does anyone have any suggestions as to how I could debug this within Kamailio? It sounds like Kamailio may be sending some incorrect packet information but I am unsure at this point. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd -- Peter Dunkley Technical Director Crocodile RCS Ltd
Re: [SR-Users] Websockets WSS problem with NOTIFY
The RSV1 bit (which is the compressed bit) should be the second bit from the left in the WebSocket frame. The first bit is the FIN (should always be one here), then you have RSV1, RSV2, and RSV3, and the last nibble of the first byte will be the opcode. Regards, Peter On 24 Jan 2013, at 14:47, Pete Kelly pke...@gmail.com wrote: Chrome 26, 24 and Firefox nightly all exhibit the same behaviour. I've decrypted the packets in wireshark, could you point me at what I am looking for to see the compressed bit? Wireshark reports (on what seems to be the problematic frame) This frame ACKs a segment we have not seen On 24 January 2013 13:50, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: Have you checked to see if there are any known bugs in the browser you are using? As the WebSocket message compression stuff is still draft the browser implementation probably won't be complete or fully tested yet. As I said, the Kamailio WebSocket implementation does not support any extensions and all the reserved bits are 0'd. So I don't think it is likely that the compressed bit is set to 1 at all. The only other thing I can suggest is capturing your TLS traffic with WireShark and importing the certificates into it so you can decode the packets. At that point you should be able to look at the binary of the frame and see if the compressed bit is set or not. Regards, Peter On Thu, 2013-01-24 at 13:45 +, Pete Kelly wrote: Hi Peter I can confirm it works correctly for WS and not WSS, and it appears to be only the NOTIFY request in the direction of Kamailio UAC. INVITE requests in the direction of Kamailio UAC are fine. I've tried it with the tls tls_disable_compression flag set to both 0 and 1 Pete On 24 January 2013 09:53, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: Hi, I've done some checking online and in the code. The compressed bit is defined in draft-ietf-hybi-permessage-compression and uses the RSV1 bit from the WebSocket frame header. As per RFC 6455 the Kamailio WebSocket implementation is careful to leave RSV1, RSV2, and RSV3 with values of 0. As this part of the code is identical for WS and WSS connections can you confirm that it works correctly for WS? Regards, Peter On Thu, 2013-01-24 at 09:09 +, Peter Dunkley wrote: I shod also add that the Kamailio WebSocket implementation does not support any extensions. So unless the deflate frame extension is implicit for TLS it will not be negotiated. Further, the implementation does not set any compressed bits and all unused flags etc should be zeroed automatically - but I will look at the code later. Peter On 24 Jan 2013, at 09:05, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: I am not sure how to investigate this. It sounds like it might be a TLS related problem (or a WebSocket/TLS interworking problem in Kamailio). I don't know anything about the Kamailio TLS implementation - I just drop WebSocket frames into it as required. I did do (a little) WSS testing and saw no problems myself. Regards, Peter On 23 Jan 2013, at 22:12, Pete Kelly pke...@gmail.com wrote: Hi, I am having an issue at the moment with SIP NOTIFY messages being sent from Kamailio (latest git master) over wss transport I am getting reports from the receiving end saying Compressed bit must be 0 if no negotiated deflate-frame extension The only reference I can find to it is at the following URL... where the problem was caused by the server miscalculating the size of the msg: http://stackoverflow.com/questions/12308728/compressed-bit-must-be-0-when-sending-a-message-to-websocket-client Does anyone have any suggestions as to how I could debug this within Kamailio? It sounds like Kamailio may be sending some incorrect packet information but I am unsure at this point. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websockets WSS problem with NOTIFY
OK. This sounds like the NOTIFY is not being routed through the WebSocket module then. Instead it is coming out as a raw SIP message. This would explain a lot. This could well be caused by the routing within Kamailio not being quite right. For example, if the ;transport=ws parameter is missing from some Route/Record-Route/Contact/Request -URI you could see something like this. It could be a code problem or just a problem with the configuration file that is causing this. I suspect it may also be related to the use of the nathelper stuff and the contact aliasing that needs to be used with WebSocket (unless you are using the latest code and have configured outbound). Regards, Peter On Thu, 2013-01-24 at 15:29 +, Pete Kelly wrote: Hi Peter, thanks for that info. It looks like all the packets marked Websocket in Wireshark are coming across OK from Kamailio. The first nibble is always 1000 as expected. However I notice now that whenever a NOTIFY is sent out from Kamailio the packet is *not* a Websocket packet, it's identified as HTTP within Wireshark and does not contain the 4 header bytes that Websocket packets seem to contain. As a result the first byte for the NOTIFY is the letter 'N' represented as 01001110. So the browser could be reading the second bit as 1, and interpreting that as meaning the compressed bit set to 1. Does that sound plausible? On 24 January 2013 14:54, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: The RSV1 bit (which is the compressed bit) should be the second bit from the left in the WebSocket frame. The first bit is the FIN (should always be one here), then you have RSV1, RSV2, and RSV3, and the last nibble of the first byte will be the opcode. Regards, Peter On 24 Jan 2013, at 14:47, Pete Kelly pke...@gmail.com wrote: Chrome 26, 24 and Firefox nightly all exhibit the same behaviour. I've decrypted the packets in wireshark, could you point me at what I am looking for to see the compressed bit? Wireshark reports (on what seems to be the problematic frame) This frame ACKs a segment we have not seen On 24 January 2013 13:50, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: Have you checked to see if there are any known bugs in the browser you are using? As the WebSocket message compression stuff is still draft the browser implementation probably won't be complete or fully tested yet. As I said, the Kamailio WebSocket implementation does not support any extensions and all the reserved bits are 0'd. So I don't think it is likely that the compressed bit is set to 1 at all. The only other thing I can suggest is capturing your TLS traffic with WireShark and importing the certificates into it so you can decode the packets. At that point you should be able to look at the binary of the frame and see if the compressed bit is set or not. Regards, Peter On Thu, 2013-01-24 at 13:45 +, Pete Kelly wrote: Hi Peter I can confirm it works correctly for WS and not WSS, and it appears to be only the NOTIFY request in the direction of Kamailio UAC. INVITE requests in the direction of Kamailio UAC are fine. I've tried it with the tls tls_disable_compression flag set to both 0 and 1 Pete On 24 January 2013 09:53, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: Hi, I've done some checking online and in the code. The compressed bit is defined in draft-ietf-hybi-permessage-compression and uses the RSV1 bit from the WebSocket frame header. As per RFC 6455 the Kamailio WebSocket implementation
Re: [SR-Users] Websockets WSS problem with NOTIFY
OK. It looks like you have a bug in the client SIP over WebSocket stack. ;transport=wss (as you have on the R-URI) is not correct. It should be ;transport=ws whether it is WS or WSS. The R-URI in the NOTIFY will be the contact that the client stack put into the SUBSCRIBE - so this wss is probably coming from the client stack. See draft-ietf-sipcore-sip-websocket section 5.2.2. The contents of the domain part of the R-URI here is also unusual - the draft recommends a made-up .invalid domain - again see draft-ietf-sipcore-sip-websocket section A.1. Also, this NOTIFY R-URI contains no ;alias= parameter - so unless you are using the latest git master and have enabled outbound you will probably have some routing problems with this. Basically, you should use the nathelper module and call add_contact_alias() for all dialog-forming and re-targetting requests (INVITE, NOTIFY, SUBSCRIBE, and UPDATE) that you receive from a WebSocket client. Then you should call handle_ruri_alias() for all requests that destined for a WebSocket client. Regards, Peter On Thu, 2013-01-24 at 15:46 +, Pete Kelly wrote: This is the ruri: NOTIFY sips:pete@10.15.20.113:55536;rtcweb-breaker=no;transport=wss SIP/2.0\r\n There is only one Via header: Via: SIP/2.0/TLS 10.15.20.170:443;branch=z9hG4bK8455.12ffc4c6.0\r\n And the Contact: Contact: sip:10.15.20.170:443;transport=ws\r\n Contact looks suspicious as ws instead of wss? Does Kamailio use the usrloc info from the REGISTER to send out a NOTIFY? On 24 January 2013 15:34, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: OK. This sounds like the NOTIFY is not being routed through the WebSocket module then. Instead it is coming out as a raw SIP message. This would explain a lot. This could well be caused by the routing within Kamailio not being quite right. For example, if the ;transport=ws parameter is missing from some Route/Record-Route/Contact/Request -URI you could see something like this. It could be a code problem or just a problem with the configuration file that is causing this. I suspect it may also be related to the use of the nathelper stuff and the contact aliasing that needs to be used with WebSocket (unless you are using the latest code and have configured outbound). Regards, Peter On Thu, 2013-01-24 at 15:29 +, Pete Kelly wrote: Hi Peter, thanks for that info. It looks like all the packets marked Websocket in Wireshark are coming across OK from Kamailio. The first nibble is always 1000 as expected. However I notice now that whenever a NOTIFY is sent out from Kamailio the packet is *not* a Websocket packet, it's identified as HTTP within Wireshark and does not contain the 4 header bytes that Websocket packets seem to contain. As a result the first byte for the NOTIFY is the letter 'N' represented as 01001110. So the browser could be reading the second bit as 1, and interpreting that as meaning the compressed bit set to 1. Does that sound plausible? On 24 January 2013 14:54, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: The RSV1 bit (which is the compressed bit) should be the second bit from the left in the WebSocket frame. The first bit is the FIN (should always be one here), then you have RSV1, RSV2, and RSV3, and the last nibble of the first byte will be the opcode. Regards, Peter On 24 Jan 2013, at 14:47, Pete Kelly pke...@gmail.com wrote: Chrome 26, 24 and Firefox nightly all exhibit the same behaviour. I've decrypted the packets in wireshark, could you point me at what I am looking for to see the compressed bit? Wireshark reports (on what seems to be the problematic frame) This frame ACKs a segment we have not seen On 24 January 2013 13:50, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote
Re: [SR-Users] Websockets WSS problem with NOTIFY
Maybe Kamailio could report an error in the logs when the unrecognised transport type is submitted? That could be handy. I am not sure how/where to put something like this though. Interesting, I had those routing problems initially, so I added the add_contact_alias() to my script but only if if (nat_uac_test(64)) passes. I'll take a look at what is happening here. It just occurred to me that as you are passed the point that handle_ruri() alias is called so you wouldn't see this in the request outside Kamailio. So please ignore my comments on this part. I am using the latest 4.0.0 sources, so I guess I could also switch to outbound. That's probably a good idea as long as you have separate edge-proxies and registrars (always a good idea to begin with). Outbound is the recommended method for SIP over WebSocket. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] PATH advertised
Hi Olle, You shouldn't need to add a double Path: as the Path: only needs to build up a unidirectional route-set from the registrar back to the terminating client - so the internal Kamailio address should be fine. The within-dialog route-set (which does need double entries) will be build from double-RRs added to the dialog forming request. One thing worth noting is that with the new outbound module included Kamailio won't even double-RR when outbound is being used (and it can be forced even for a proxy that isn't at the edge) as the flow-token contains all of the information that would be in the extra RR (and of course Path: contents for outbound are the same). Regards, Peter Hi! If I'm running Kamailio behind NAT and need to add a Path header with the outside IP address and port, that is configured as the advertise address - how do I do that? Can I add two Path headers at the same time? I can't find a way to add it like I can with Record-Route headers. /O ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] FOSDEM
Hello, Are there any plans for a Kamailio meal/get together on the Saturday evening this year? Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] FOSDEM
All the guys from Crocodile will be at the Hotel du Congres. Stayed there last year, it was basic but clean and central. We will be arriving Saturday afternoon. Peter On 9 Jan 2013, at 21:55, Klaus Darilion klaus.mailingli...@pernau.at wrote: Am 09.01.2013 18:13, schrieb Peter Dunkley: Hello, Are there any plans for a Kamailio meal/get together on the Saturday evening this year? Peter ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev Is there a hotel suggestion for the VoIP guys? regards Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] Testing SIP Outbound / RFC5626 (Was: Freezing for next major release)
Richard Brady writes: i didn't find in rfc5626 a requirement that registrar should remove 430 flow contact, Closest I can find is: EP1 no longer has a flow to Bob, so it responds with a 430 (Flow Failed) response. The proxy removes the stale registration and tries the next binding for the same instance. it is thus the job of the proxy, not the registrar, to remove the currently unusable registration. the proxy could do it either by sending un-REGISTER request to registrar or via an mi/rpc command. the former is possible now. the latter requires that instance and reg_id params are added to ul_rm_contact mi command. When we were discussing implementing outbound someone (I think it was Inaki) did mention that some parts of the outbound spec were unclear on precisely how to implement things. This could well be one of those parts - no mention except in an example, and even there the description feels a bit wrong. I just don't like the idea that an edge proxy should need to be a UAC and generate an un-REGISTER. Adding an MI/RPC command would be implementation specific. Making the registrar capable of removing a contact in response to receiving a 430 seems far more logical and is going to be no harder than adding an MI/RPC command to do it. I will add a new exported function to registrar, probably called unregister_contact(), at some point in the future. I don't think it is a big job but it almost certainly won't happen before the freeze - it may be a couple of weeks before I have time to come back to it. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Testing SIP Outbound / RFC5626 (Was: Freezing for next major release)
Hi Richard, I haven't been directly involved with the coding of the registrar stuff. This does sound like testing worth doing. Also worth adding tests to check that if you get a 430 back after routing to one reg-id you can try the next. Peter On Mon, 2013-01-07 at 12:54 +, Richard Brady wrote: Hi Peter Great work on this! We'd like to help you test. The test I have in mind, which we could create using SIPp, would be to register multiple contacts with the same instance-id (i.e. sip.instance param) but different reg-id params. Then send an INVITE to that AoR and make sure the forking is only per instance-id and not per reg-id. This could be repeated in multiple permutations of instance-ids and reg-ids. This would be a test of save() and lookup() more than anything else. Is that what you had in mind? Richard On 3 January 2013 14:13, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: I hope to get the outbound edge proxy and flow timer stuff into master by Monday, but it could really do with some additional testing especially in conjunction with the reg-id stuff in registrar/usrloc (which I have no idea how to use). Peter ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] Testing SIP Outbound / RFC5626 (Was: Freezing for next major release)
Hi Juha, I assumed as much, but what I haven't had the chance to do myself yet is the full end-to-end scenario where the Kamailio edge proxy indicates a broken flow and the registrar successfully re-routes through another flow. Peter On 7 Jan 2013, at 19:46, Juha Heinanen j...@tutpro.com wrote: Peter Dunkley writes: I haven't been directly involved with the coding of the registrar stuff. This does sound like testing worth doing. i did the registrar stuff and tested it with baresip, which supports outbound. more testing is of course welcome. -- juha ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] Testing SIP Outbound / RFC5626 (Was: Freezing for next major release)
Hi Juha, I was wondering if you could provide an example of how to use lookup()/lookup_branches(), t_load_contacts(), t_next_contacts(), and t_next_contacts_flows() for registrar supporting outbound with an edge proxy? From the documentation of these functions it does look like parallel and serial forking and outbound should just work if they are used properly. Is that correct? One requirement of an outbound capable registrar is that if a flow fails (edge proxy returns a 430) the registrar should realise that the flow is now dead and remove that contact binding from its database so it is not used again as well as trying the next contact. I can't see anything that will do this? Is this missing? Thanks, Peter Peter Dunkley writes: I haven't been directly involved with the coding of the registrar stuff. This does sound like testing worth doing. i did the registrar stuff and tested it with baresip, which supports outbound. more testing is of course welcome. -- juha ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] status of kamailio stun server?
I haven't done any testing of it, but supporting STUN on the same port as SIP is a requirement for an edge proxy supporting outbound. So I don't think it should be obsoleted or removed. Peter On 15 Dec 2012, at 17:49, Juha Heinanen j...@tutpro.com wrote: i tried to use kamailio built-in stun server, but failed, because it supports only binding requests. my ua tried to send it allocate request lifetime request, which failed with 600 global failure response. what is the status of kamailio's stun server? is anyone interested in developing it further or should we obsolete the server and start using some external server? -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Help
Hi Roy, Please keep emails on list. Regards, Peter On Fri, 2012-12-14 at 09:31 -0800, Raj Roy Ghandhi wrote: Dear Peter, Thanks for all the support that you gave me during the Kamalio configuration with RTPProxy. Currently it works fine with IP Phones but not soft clients. (Jitsi) If I give you my server information would you be able to check that out for me. I was lost for that issue. Best Regards, Roy. -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] OT: MSRP over WebSocket Javascript stack
Hi Daniel, You do need a SIP stack to use this. We are using JsSIP here at Crocodile. This has only been tested with Google Chrome so far, but there shouldn't be anything browser specific in the stack. Regards, Peter On 14 Dec 2012, at 19:59, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 12/13/12 11:07 PM, Peter Dunkley wrote: Hi, Crocodile has just open-sourced our MSRP over WebSocket (see http://tools.ietf.org/html/draft-pd-msrp-websocket) Javascript stack. The project is hosted on Google Code: http://code.google.com/p/crocodile-msrp/ The stack is distributed using the MIT License and was developed and tested along side the Kamailio MSRP over WebSocket implementation. thanks for sharing it and your extensive testing and contributions to msrp module in kamailio! It is pure msrp, right? Meaning it has to be used on top of a sip stack. What browsers have been used for testing? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] OT: MSRP over WebSocket Javascript stack
Hi, Crocodile has just open-sourced our MSRP over WebSocket (see http://tools.ietf.org/html/draft-pd-msrp-websocket) Javascript stack. The project is hosted on Google Code: http://code.google.com/p/crocodile-msrp/ The stack is distributed using the MIT License and was developed and tested along side the Kamailio MSRP over WebSocket implementation. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websocket with Kamailio 3.2.2 release
I just tried jssip in Chrome with Asterisk (directly and via Kamailio): signaling work (no intensive testing) but audio does not work due to a bug in Chrome. I also tried Opera 12.11 and Firefox nightly 2012-11-27 but it sems that both do not support webrtc at all. Seems like we can only test Chrome vs. Chrome. WebRTC is in Firefox nightly. I haven't tried it myself because the APIs have slightly different names (start Moz instead of Webkit). This means that WebRTC software needs to detect the browser type and call differently named functions. Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Call from SipML5 - PSTN
Unless your PSTN gateway supports the RTP/SAVPF media profile - I don't know of any that do - this will not work. Regards, Peter Hi, From all of your support now I can call from 1. IP Phone -- IP Phone 2. Web Page -- Web Page 3. IP Phone - PSTN without any issue But when I try to call from Web Page to PSTN then it tries to call sip:00xx89...@mysipdomain.com and that time out. Trying to figure out how to get this work ? Can anybody guide me on this please. Best Regards, Roy. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Call from SipML5 - PSTN
You need some sort of media gateway or server. I don't know of any that currently support this. Peter Raj Roy Ghandhi roy.gan...@gmail.com wrote: ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websocket with Kamailio 3.2.2 release
We can also use jssip library. They have some demo to try. That won't fix his testing with non-WebSocket/browser client problems. Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websocket with Kamailio 3.2.2 release
Hello, This issue has come up a lot before. SIP over WebSocket is a signalling protocol, the SDP (carried in the SIP requests) describes the media profile/capabilities of each user agent. In the case of a WebSocket connection from a browser (using WebRTC for media) the media profile will be RTP/SAVPF. However, most clients do not support this profile, most clients just support RTP/AVP. This means that, while SIP between your browser and client through Kamailio will work (and is working here), you are not going to be able to establish media sessions unless the client supports RTP/SAVPF. The only clients that I know of which support RTP/SAVPF are the ones from Doubango. Regards, Peter Hi, Thanks a lot for the support. Now I can do calls. NAT traversal and Web Socket are working fine. Followings are my test scenarios :- 1. UTStarCOM IP phone - UTStarCOM IP phone - working fine 2. WebPage -- web Page :- working fine 3. UTStarCOM IP phone -- Web Page :- *ERROR *Failed to get local SDP offer when try to answer 4 Web Page UTStarCOM IP phone :- *ERROR * Request Time out 5. Jitsi - Jitsi :- No voice path. Tested with most available soft phones and all are same. ** Kamalio is running on public dedicated server. All the tested clients are behind NAT Please help me out to figure the issue. I want to make the Kamailio to work with all above scenario. Best Regards, Roy On Fri, Nov 23, 2012 at 4:35 PM, Raj Roy Ghandhi roy.gan...@gmail.comwrote: Hi, GOOOD NEWS :-) I was able to compile and use Kamailio 3.4 with web-socket. Working well with SIPML5. Will be testing more and update you if I got into issue. Best Regards, Roy On Fri, Nov 23, 2012 at 12:46 AM, Raj Roy Ghandhi roy.gan...@gmail.comwrote: Hi, Thanks a lot for the support. Now I am on track. :-) Will update you if I fall into trouble. Best Regards, Roy On Thu, Nov 22, 2012 at 3:39 PM, Jesús Pérez Rubio jesus.pe...@quobis.com wrote: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git:) 2012/11/22 Raj Roy Ghandhi roy.gan...@gmail.com Hi, Thanks for the info. Highly appreciated. I like to stay with Kamailio. Could you please provide me valid URL to download Kamailio 3.4. I could not find it. Best Regards, Roy. On Thu, Nov 22, 2012 at 3:11 PM, Peter Dunkley peter.dunk...@crocodile-rcs.com wrote: ** Yes. But make sure the WS server supports the Path extension (OverSIP does) and you enable Path support in the Kamailio registrar module. Alternatively, you could just build and use Kamailio 3.4. On Thu, 2012-11-22 at 14:19 +0530, Raj Roy Ghandhi wrote: Hi, Thanks for the reply. Can I use OverSIP or any other web socket server for the web-socket layer until Kamailio 3.4 comes. I need to make the web-phone to communicate with my Kamailio version 3.3. Best Regards, Roy. On Wed, Nov 21, 2012 at 10:28 PM, Jesús Pérez Rubio jesus.pe...@quobis.com wrote: Hi, Websockets module is only availiable in devel version (3.4) at this moment, you should try with it. http://www.kamailio.org/wiki/features/new-in-devel#websocket - A Kamailio.cfg example with websockets and MYSQL support: https://gist.github.com/4066845 2012/11/21 Raj Roy Ghandhi roy.gan...@gmail.com Hi, I am trying to integrate the websocket module into release 3.2.2. But I am unable to do that. Please guide me to get it done. Regards, Roy. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com ___ SIP Express Router (SER) and Kamailio (OpenSER
Re: [SR-Users] Websocket with Kamailio 3.2.2 release
Yes. But make sure the WS server supports the Path extension (OverSIP does) and you enable Path support in the Kamailio registrar module. Alternatively, you could just build and use Kamailio 3.4. On Thu, 2012-11-22 at 14:19 +0530, Raj Roy Ghandhi wrote: Hi, Thanks for the reply. Can I use OverSIP or any other web socket server for the web-socket layer until Kamailio 3.4 comes. I need to make the web-phone to communicate with my Kamailio version 3.3. Best Regards, Roy. On Wed, Nov 21, 2012 at 10:28 PM, Jesús Pérez Rubio jesus.pe...@quobis.com wrote: Hi, Websockets module is only availiable in devel version (3.4) at this moment, you should try with it. http://www.kamailio.org/wiki/features/new-in-devel#websocket - A Kamailio.cfg example with websockets and MYSQL support: https://gist.github.com/4066845 2012/11/21 Raj Roy Ghandhi roy.gan...@gmail.com Hi, I am trying to integrate the websocket module into release 3.2.2. But I am unable to do that. Please guide me to get it done. Regards, Roy. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Websocket with Kamailio 3.2.2 release
Hi, The websocket module requires several hundred lines of changes in Kamailio core and other modules to work, you cannot just copy it into Kamailio 3.2 and use it (nor will you be able to copy it into Kamailio 3.3). Regards, Peter On Wed, 2012-11-21 at 20:50 +0530, Raj Roy Ghandhi wrote: Hi, I am trying to integrate the websocket module into release 3.2.2. But I am unable to do that. Please guide me to get it done. Regards, Roy. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] Next IRC devel meeting
Friday works for me. On Mon, 2012-11-19 at 09:40 +0100, Daniel-Constantin Mierla wrote: Hello, would this week Friday at same time suit more people? I created a page to collect the topics to discuss, feel free to add content there: * http://www.kamailio.org/wiki/devel/irc-meetings/2012b Cheers, Daniel On 11/14/12 7:46 PM, Peter Dunkley wrote: Hi, I would like to be involved, but am not available on Monday or Tuesday. I am currently available any time on Wednesday, Thursday, and Friday. Regards, Peter On Wed, 2012-11-14 at 13:47 +0100, Olle E. Johansson wrote: 14 nov 2012 kl. 13:23 skrev Daniel-Constantin Mierla mico...@gmail.com: Hello, I am thinking of having our next IRC devel meeting soon, to plan the next major release and review current stable releases and the environment around the project (e.g., if you can add anything else to the project to make life easier for developers and users). Great! I'm proposing next week, Thursday, at 15:00GMT, on Freenode IRC server, channel #sip-router. Anyone able to join? If it is not a convenient date for you, just propose alternatives and we will select the one that meets the constraints of the most developers and users willing to participate. I will be travelling between a hotel and an airport abroad then... Monday or Tuesday that week would be better for me. My participation can't be considered critical though, as I'm more focusing on cleaning up docs for the new release. A question that it's about time to ask is what a major release - like a 4.0 is compared with the differences between 3.3 and a possible 3.4. With the additions of websockets and MSRP and potentially SIP outbound plus much more the coming release is not insignificant. Cheers /O ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Peter Dunkley Technical Director Crocodile RCS Ltd -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] Next IRC devel meeting
Friday this week works for me, but Friday next week I will be travelling. Peter On Tue, 2012-11-20 at 10:53 +0100, Carsten Bock wrote: Next week Friday works for me better, too. Carsten 2012/11/20 Peter Dunkley peter.dunk...@crocodile-rcs.com Friday works for me. On Mon, 2012-11-19 at 09:40 +0100, Daniel-Constantin Mierla wrote: Hello, would this week Friday at same time suit more people? I created a page to collect the topics to discuss, feel free to add content there: * http://www.kamailio.org/wiki/devel/irc-meetings/2012b Cheers, Daniel On 11/14/12 7:46 PM, Peter Dunkley wrote: Hi, I would like to be involved, but am not available on Monday or Tuesday. I am currently available any time on Wednesday, Thursday, and Friday. Regards, Peter On Wed, 2012-11-14 at 13:47 +0100, Olle E. Johansson wrote: 14 nov 2012 kl. 13:23 skrev Daniel-Constantin Mierla mico...@gmail.com: Hello, I am thinking of having our next IRC devel meeting soon, to plan the next major release and review current stable releases and the environment around the project (e.g., if you can add anything else to the project to make life easier for developers and users). Great! I'm proposing next week, Thursday, at 15:00GMT, on Freenode IRC server, channel #sip-router. Anyone able to join? If it is not a convenient date for you, just propose alternatives and we will select the one that meets the constraints of the most developers and users willing to participate. I will be travelling between a hotel and an airport abroad then... Monday or Tuesday that week would be better for me. My participation can't be considered critical though, as I'm more focusing on cleaning up docs for the new release. A question that it's about time to ask is what a major release - like a 4.0 is compared with the differences between 3.3 and a possible 3.4. With the additions of websockets and MSRP and potentially SIP outbound plus much more the coming release is not insignificant. Cheers /O ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Peter Dunkley Technical Director Crocodile RCS Ltd -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:cars...@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] Next IRC devel meeting
Hi, I would like to be involved, but am not available on Monday or Tuesday. I am currently available any time on Wednesday, Thursday, and Friday. Regards, Peter On Wed, 2012-11-14 at 13:47 +0100, Olle E. Johansson wrote: 14 nov 2012 kl. 13:23 skrev Daniel-Constantin Mierla mico...@gmail.com: Hello, I am thinking of having our next IRC devel meeting soon, to plan the next major release and review current stable releases and the environment around the project (e.g., if you can add anything else to the project to make life easier for developers and users). Great! I'm proposing next week, Thursday, at 15:00GMT, on Freenode IRC server, channel #sip-router. Anyone able to join? If it is not a convenient date for you, just propose alternatives and we will select the one that meets the constraints of the most developers and users willing to participate. I will be travelling between a hotel and an airport abroad then... Monday or Tuesday that week would be better for me. My participation can't be considered critical though, as I'm more focusing on cleaning up docs for the new release. A question that it's about time to ask is what a major release - like a 4.0 is compared with the differences between 3.3 and a possible 3.4. With the additions of websockets and MSRP and potentially SIP outbound plus much more the coming release is not insignificant. Cheers /O ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [sr-dev] presence module makes excessive cleanups when subs_db_mode=3
This is by design. When you have many thousands of presence subscriptions you need to service them evenly across time instead of in big lumps. If you don't want this behaviour then you can set the notifier_processes modparam to 0. But, if you do this you should consider not using subs_db_mode 3 as there are many race-hazards in presence that are fixed by using the notifier_processes. Regards, Peter On Fri, 2012-11-09 at 12:01 +0100, Andrew Pogrebennyk wrote: Hi, when presence module is running with subs_db_mode=3 it makes an excessive number of SQL select queries, litelly dozens per second: 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event'presence.winfo' 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event'presence.winfo' 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=116 AND event='presence.winfo' 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=64 AND event='presence.winfo' 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event'presence.winfo' 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=14 AND event='presence.winfo' 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event'presence.winfo' 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event'presence.winfo' 25233 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=118 AND event='presence.winfo' 25232 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=66 AND event='presence.winfo' 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event'presence.winfo' 25231 Query select presentity_uri,callid,to_tag,from_tag,event from active_watchers where updated=16 AND event='presence.winfo' Steps to reproduce: registers two subscribers in jitsi and add them to contact lists of each other. The kamailio version is 3.3.2. I'm not doing anything special configuration-wise: loadmodule presence.so modparam(presence, db_url, mysql://kamailio:snbF93@localhost/kamailio) # in 3.3 the fallback2db change to subs_db_mode modparam(presence, subs_db_mode, 3) modparam(presence, notifier_processes, 3) loadmodule presence_xml.so modparam(presence_xml, db_url, mysql://kamailio:snbF93@localhost/kamailio) modparam(presence_xml, force_active, 0) modparam(presence_xml, integrated_xcap_server, 1) # retry-after 5 minutes modparam(presence_xml, xcapauth_userdel_reason, probation;retry-after=300) Q: it is a bug of a feature? :) Thanks. Andrew ___ sr-dev mailing list sr-...@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Peter Dunkley Technical Director Crocodile RCS Ltd ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] ERROR: presence not working correctly with Kamailio 3.3.2-1.1 and Postgresql 9.1
Try using the same version of PostgreSQL client library and server.
I believe the default handling of binary blobs changed between PostgreSQL
8.x and 9.x. This could well explain your problem.
Kamailio works for me with PostgreSQL 9.0 client library and server.
Regards,
Peter
Hi All,
I've got a problem with Kamailio 3.3.2-1.1 on CentOS 6.3 with locally
postgresql 8.4.13-1.el6_3 libraries, connected to a remote
postgresql91-9.1.6-1PGDP.rhel6 server
The presence module tries to insert a record in the presentity table,
with some xml in the body column.
But the body value in the postgresql table is not represented as a
string '?xml ...etc' But encodes as
'\x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462...etc'
This gives problems when the presense module is trying to send out a
notify based on the value in the database.
With debugging, this gives the error:
Entity: line 1: parser error : Start tag expected, '' not found
x3c3f786d6c2076657273696f6e3d27312e302720656e636f64696e673d275554462d38273f3e3c
The \x in front of the hex encodes string is not properly interpreted by
postgresql or kamailio.
The body column of the table presentity is a bytea column, and I believe
postgresql 9 outputs these bytea columns a bit differently then
postgresql 8.
When I try to run the sql statement:
insert into presentity
(domain,username,event,etag,sender,body,received_time,expires ) values
('newsip.lifexs.nl','00086','presence','a.1352107949.18632.27.0','','?xml
version=''1.0'' encoding=''UTF-',1343534532,1345213723)
On the postgresql 9.1 server locally, I still get the '\x3c3f786d6c...
etc' value in the database.
The bytea_output setting in postgresql 9 is now standard set to 'hex' in
stead of escape.
I've changed the bytea_output setting in postgresql.conf to 'escape',
and then the presence is working without any issues.
So maybe the database module of postgresql has to be changed to pick up
these bytea encodings properly?
Because i think kamailio should work correctly with the default
postgresql settings.
With kind regards,
Robert Verspuy
--
*Exa-Omicron*
Eenspan 8-K
3897 AL Zeewolde
http://exa.nl
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