[SR-Users] Web softphone using websockets against Kamailio

[Pirjo Ahvenainen]

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

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Re: [SR-Users] Is kamailio creating multiple connections to db?

[Daniel-Constantin Mierla]

Adding references to two solutions:
- shared IP with cross replication between mysql servers
- mysql cluster

Cheers,
Daniel

On 1/25/13 8:18 PM, Rumen Mihailov wrote:


Hello Mino,

Sorry for the offtopic, what HA database solution do you use ?

Regards,
Rumen

On 25 Jan 2013 16:22, "Mino Haluz" > wrote:


Hi,

I made some DB test on our HA cluster, and I got that for 1
process I have 1000 queries/sec and for 11 processes 3000
queries/sec. That's why I would to ask if every kamailio process
that is spawned has its own database connection or queries are put
in the queue and executed by some main process ? I would like to
know this, so that I could assess maximum call count/sec that db
can handle.

Thank you,
Mino

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Re: [SR-Users] Generating logging based on SIP messages

[Daniel-Constantin Mierla]

Hello,

siptrace module can store directly to a database. It stores some 
attributes separately, also it can selectively do it only for some 
requests based on config file. Looking over that part of code in 
siptrace can be a starting point.


Cheers,
Daniel

On 1/24/13 12:41 PM, Grant Bagdasarian wrote:


Hello,

A while ago I came across the siptrace and sipcapture modules and 
thought this would be a good way to generate logging out of all the 
SIP messages that are passed through Kamailio and stored in the 
siptrace/sipcapture table.


My idea was to extract certain values from SIP messages and store 
these in our database. Also each SIP Request or Response would be 
mapped to a certain status so we would be able to monitor which call 
is in which state.


I still have a lot of work to do to think this all out, since it's 
still only an idea. I'm just wondering if anyone could give me some 
pointers into what to keep in mind when doing this.


Regards,

Grant



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Re: [SR-Users] Problem with 4.0 nightlies and IMS - sem_post undefined

[Daniel-Constantin Mierla]

Hello,

do you still get the issues? The ims modules were not touched lately. 
sem_post is from the core.


Cheers,
Daniel

On 1/22/13 2:19 PM, Barry Flanagan wrote:

Hi

I am trying to run the Ubuntu nightlies from 
http://deb.kamailio.org/kamailiodev-nightly and loading the p-cscf 
config files.


Things were working fine while I was on the 20130110 nightlies, but 
when I upgraded to 20130122 I get the following errors. Any have an 
idea why sem_post is undefined and how to fix this?




 0(1980) ERROR:  [sr_module.c:572]: ERROR: load_module: could 
not open module : 
/usr/lib64/kamailio/modules/ims_registrar_pcscf.so: undefined symbol: 
sem_post
 0(1980) ERROR:  [sr_module.c:572]: ERROR: load_module: could 
not open module : 
/usr/lib64/kamailio/modules/cdp.so: undefined symbol: sem_post
 0(1980) ERROR:  [sr_module.c:572]: ERROR: load_module: could 
not open module : 
/usr/lib64/kamailio/modules/ims_qos.so: undefined symbol: sem_post


Thanks

-Barry Flanagan







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Re: [SR-Users] Pseudo variable to get instance in the contact....

[Daniel-Constantin Mierla]

Hello,

On 1/21/13 1:35 PM, Krishna Kurapati wrote:

Hi,

I added in my local branch "$ct.inst" pseudo variable to retrieve 
instance. Since instance could be an important attribute to make 
routing decisions, having quick access to it in the message may be useful.


Anyone else need this functionality?
I would go for a class of PVs, so attributes can be accessed as 
$contact(attr) - for example $contact(inst), $contact(uri) ... it would 
be more standard and easier to track and to extend.


Cheers,
Daniel

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Re: [SR-Users] Pseudo variable to access SIP instance in contact

[Daniel-Constantin Mierla]

Hello,

On 1/19/13 4:24 PM, Krishna Kurapati wrote:

Hi,

Is there a way to directly access the SIP instance present in message 
from the script?
not a dedicated variable now, but you can get along with 'to' or 
'namaddr' and 'param' transformations.


The alternative would be via selects, try:

http://www.kamailio.org/wiki/cookbooks/devel/selects#contacturiparams_s

Cheers,
Daniel

--
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Re: [SR-Users] CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg - Kamailio 3.3.x

[Daniel-Constantin Mierla]

Hello,

the log message is not that critical as in the message. It should be a 
warning.


Can you provide more details about running environment? Was it under 
stress test or just normal running? How many active calls were around? 
Could you list the dialogs and see what is the state for such dialog?


Cheers,
Daniel

On 1/18/13 2:11 PM, Rinor Hoxha wrote:

Hi list,

I'm having some issues with dialog module. From time to time I get the 
following error:


ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil)


[root@proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" 
| sort | uniq -c


  1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil)
  1 Jan 17 14:39:30 proxy /usr/local/kamailio/sbin/kamailio[4100]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil)
  1 Jan 17 15:06:21 proxy /usr/local/kamailio/sbin/kamailio[4100]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil)
  1 Jan 17 19:31:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil)
  1 Jan 17 19:31:42 proxy /usr/local/kamailio/sbin/kamailio[4100]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil)
  1 Jan 17 20:05:32 proxy /usr/local/kamailio/sbin/kamailio[4100]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil)


This one was flooding my log file: tl=0x2aeeea1c6840

  24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil)
   5709 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4100]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil)


  61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil)
  22812 Jan 17 21:54:01 proxy /usr/local/kamailio/sbin/kamailio[4102]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil)
  21137 Jan 17 21:54:02 proxy /usr/local/kamailio/sbin/kamailio[4102]: 
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg 
tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil)


where pid 4100 is slow timer and pid 4102 is timer. Any idea why is 
this happening and how to fix it.
(I suspect that may be some issues with our server memory...checking 
now...and will inform). However wanted to raise this issue in case 
anyone else is having the same.


Thanks, Rinor




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Re: [SR-Users] siremis CDR gneration procedure

[Daniel-Constantin Mierla]

Hello,

On 1/17/13 3:38 PM, Klaus Darilion wrote:

Hi!

IMO the siremis CDR generation procedure is buggy:

CREATE PROCEDURE `kamailio_cdrs`()
  DECLARE inv_cursor CURSOR FOR SELECT src_user, src_domain, dst_user,
 dst_domain, dst_ouser, time, callid,from_tag, to_tag, src_ip
 FROM acc
 where method='INVITE' and cdr_id='0';
...
  REPEAT
FETCH inv_cursor INTO v_src_user,
v_src_domain, v_dst_user, v_dst_domain,
v_dst_ouser, v_inv_time, v_callid, v_from_tag,
v_to_tag, v_src_ip;
IF NOT done THEN
  SET bye_record = 0;
  SELECT 1, time INTO bye_record, v_bye_time FROM acc WHERE
 method='BYE' AND callid=v_callid AND
 ((from_tag=v_from_tag AND to_tag=v_to_tag)
 OR (from_tag=v_to_tag AND to_tag=v_from_tag))
 ORDER BY time ASC LIMIT 1;
  IF bye_record = 1 THEN
INSERT INTO cdrs (src_username,src_domain,dst_username,
 dst_domain,dst_ousername,call_start_time,
 duration,sip_call_id,
 sip_from_tag,sip_to_tag,src_ip,created) VALUES
 (v_src_user, v_src_domain,v_dst_user,
 v_dst_domain,v_dst_ouser,v_inv_time,
UNIX_TIMESTAMP(v_bye_time)-UNIX_TIMESTAMP(v_inv_time),
 v_callid,v_from_tag,v_to_tag,v_src_ip,NOW());
UPDATE acc SET cdr_id=last_insert_id() WHERE callid=v_callid
 AND from_tag=v_from_tag AND to_tag=v_to_tag;
  END IF;
  SET done = 0;
END IF;
  UNTIL done END REPEAT;
...

1. The UPDATE query will not UPDATE BYE records which were sent from 
CALLEE to CALLER. This can be easily fixed with:


UPDATE acc SET cdr_id=last_insert_id() WHERE callid=v_callid
 AND ( (from_tag=v_from_tag AND to_tag=v_to_tag) OR
   (to_tag=v_from_tag AND from_tag=v_to_tag) );


the idea is to update the INVITE record not to select it in future 
executions.


2. (minor) It does not consider reINVITEs. I know the default Kamailio 
config does not account reINVITEs, but I think when reINVITEs are 
accounted too, this can mess up the CDR generation.


For re-invites, the accounting record has to be extended with an extra 
filed to mark the first one from the config. That should be the easiest, 
timestamp can be another one but will make the sql procedure more complex.


Cheers,
Daniel




regards
Klaus

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Re: [SR-Users] iptel x media5-fone problem

[Daniel-Constantin Mierla]

Hello,

for iptel.org sip service use the mailing list at:

http://lists.iptel.org/mailman/listinfo/services

Cheers,
Daniel

On 1/17/13 1:23 PM, Kasper Hämäläinen wrote:

hi guys,

hope u got time to help me.

Im first timer testing VoIP and SIP.

I created account to ur iptel.org  site.
I downloaded "Jitsi"-software.

First when I opened it, I login with my username and password.

Then I go to "add new account", give same login username and password,
but Jitsi says that my password is wrong..wtf.

And

I installed Media5-Fone to my iPhone 4S.

When adding SIP-account it's says again that I got wrong password.

And YES, I didn't mistype it.

Thanks

- Kasper


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Re: [SR-Users] Pseudo variable to get instance in the contact....

[Olle E. Johansson]

28 jan 2013 kl. 10:21 skrev Daniel-Constantin Mierla :

> Hello,
> 
> On 1/21/13 1:35 PM, Krishna Kurapati wrote:
>> Hi,
>> 
>> I added in my local branch "$ct.inst" pseudo variable to retrieve instance. 
>> Since instance could be an important attribute to make routing decisions, 
>> having quick access to it in the message may be useful.
>> 
>> Anyone else need this functionality?
> I would go for a class of PVs, so attributes can be accessed as 
> $contact(attr) - for example $contact(inst), $contact(uri) ... it would be 
> more standard and easier to track and to extend.

Would that be faster than using the current transformation to pick out header 
parameters?

/O
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Re: [SR-Users] Generating logging based on SIP messages

[Grant Bagdasarian]

Hello,

>From what I understand after reading the siptrace and sipcapture modules is 
>the following:


1)  My main kamailio server must use the duplicate_uri modparam to send a 
duplicate of the sip message to the sipcapture server

2)  The sipcapture server will receive these messages and store them in the 
database

3)  I will use an instead of trigger on the sipcapture table to insert 
these messages in a database queue for processing by an external service.

Although SIPtrace already stores a sip message in a database, sipcapture also 
splits each header in the message to  a separate column. This is exactly what I 
need.

Regards,

Grant


From: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel-Constantin 
Mierla
Sent: maandag 28 januari 2013 10:17
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users 
Mailing List
Subject: Re: [SR-Users] Generating logging based on SIP messages

Hello,

siptrace module can store directly to a database. It stores some attributes 
separately, also it can selectively do it only for some requests based on 
config file. Looking over that part of code in siptrace can be a starting point.

Cheers,
Daniel
On 1/24/13 12:41 PM, Grant Bagdasarian wrote:
Hello,

A while ago I came across the siptrace and sipcapture modules and thought this 
would be a good way to generate logging out of all the SIP messages that are 
passed through Kamailio and stored in the siptrace/sipcapture table.
My idea was to extract certain values from SIP messages and store these in our 
database. Also each SIP Request or Response would be mapped to a certain status 
so we would be able to monitor which call is in which state.

I still have a lot of work to do to think this all out, since it's still only 
an idea. I'm just wondering if anyone could give me some pointers into what to 
keep in mind when doing this.

Regards,

Grant






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Kamailio World Conference, April 16-17, 2013, Berlin

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Re: [SR-Users] Is kamailio creating multiple connections to db?

[Mino Haluz]

Yes, mysql cluster. It seems to be 2x slower than mysqld myisam, but who
cares, 3000q/s is ok for me.


On Mon, Jan 28, 2013 at 10:10 AM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:

>  Adding references to two solutions:
> - shared IP with cross replication between mysql servers
> - mysql cluster
>
> Cheers,
> Daniel
>
>
> On 1/25/13 8:18 PM, Rumen Mihailov wrote:
>
> Hello Mino,
>
> Sorry for the offtopic, what HA database solution do you use ?
>
> Regards,
> Rumen
> On 25 Jan 2013 16:22, "Mino Haluz"  wrote:
>
>> Hi,
>>
>>  I made some DB test on our HA cluster, and I got that for 1 process I
>> have 1000 queries/sec and for 11 processes 3000 queries/sec. That's why I
>> would to ask if every kamailio process that is spawned has its own database
>> connection or queries are put in the queue and executed by some main
>> process ? I would like to know this, so that I could assess maximum call
>> count/sec that db can handle.
>>
>>  Thank you,
>> Mino
>>
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>
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>
> --
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> - http://www.linkedin.com/in/miconda
> Kamailio World Conference, April 16-17, 2013, Berlin
>  - http://conference.kamailio.com -
>
>
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[SR-Users] event_route for pre-dialog start

[Mino Haluz]

Hi,

I think that it is more than likely not implemented, but is there any
event-route that is triggered just before event_route[dialog:start] ? I
need to check some security flags before the dialog is created. But it is
too early too check them in relay route..

Mino
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[SR-Users] Error searching $keepalive.socket pvar in NAT Traversal module

[vito]

Hello,

I started to use Kamailio few months ago. Maybe this is a stupid issue but
I can´t start Kamailio with NAT TRAVERSAL module and $keepalive.socket
pseudo-variable.
My script configuration is the default template adding this lines
(extracted from nat traversal doc):

...
loadmodule "nat_traversal.so"
...

...
$fs = $keepalive.socket($du);
...

I have this error:

-e
 0(17455) ERROR:  [pvapi.c:501]: error searching pvar "keepalive"
 0(17455) ERROR:  [pvapi.c:705]: wrong char [e/101] in [$keepalive]
at [9 (0)]
 0(17455) :  [cfg.y:3330]: parse error in config file
/etc/kamailio/kamailio.cfg, line 410, column 8-17: unknown script pseudo
variable $keepalive
ERROR: bad config file (1 errors)


I tried in this environments:

 Ubuntu 10.04.4 LTS x86_64   Kamailio 3.3.3+git20130128+lucid1
 Ubuntu 10.04.4 LTS i686 Kamailio 3.0.4+lucid1
 Ubuntu 12.04.1 LTS x86_64   Kamailio 3.3.2+precise1

All packages instaled from deb repositories (http://deb.kamailio.org/)

I think there are problem parsing "$keepalive.socket" pseudovariable,
because Kamailio doesn´t scape this character "." and it doesn´t recognise
the pseudovar.

Does anyone have this problem?


Regards
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[SR-Users] TLS certificate chain Android verification

[Nord7]

Hello all.

Our company buy Thawte SSL Web server certificate.

I'm enabled in Kamailio 3.3 TLS support and configured him.

Adding to "myhost.pem" certificate, two "intermediate" certificates (like
in chain) "cat myhost.pem SSL_SecondaryCA.pem SSL_PrimaryCA.pem >
chain-cert.pem".

Remark: I use this chain on apache2.  And Android trusted to this
certificate chain.

After this  i use chain-cert.pem with KAMAILIO 3.3 and SIP client "BRIA
Android". "Bria Android" says certificate is not trusted. Next step "BRIA
IPHONE" and certificate chain works (trusted)!

BRIA support says me - BRIA don't trusts certificate if system don't trusts.

Why Android don't trusts to my chain-certificates with KAMAILIO? If his
trust to same certificate with Apache?

I'm understand this is not Kamailio problem, but maybe any one who have
expirience in this problem help me? I know about bouncycastle, but this is
not best decision for me. Maybe something else can help?


I tryed this with Android 2.3.7, 4.0, 4.1.

Best regards Kirill.
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Re: [SR-Users] Kamailio stability/timing problem w.r.t. registrations?

[Sotas Development]

Hi Ovidiu,

Thanks for the warning! We did not yet have much success running the
current master branch, though this may well be a resource problem on
the target platform.

For the moment, we decided to switch back to openser 1.3.5 and wait
for the official 4.0 release.

Regards,
Michiel Veldkamp


On Thu, Jan 17, 2013 at 7:01 PM, Ovidiu Sas  wrote:

> If you are running the stable version, there's need for heavy Makefile
> patching in order to properly cross compile (not to include and link
> to host libs).
> The trunk has everything fixed and it's cross-compiling properly for
> most of the modules.
> Make sure that your binaries are properly cross compiled.
>
> Depending on your ARM CPU, atomic locks may or may not work.
> I tested openser without atomic locks (using regular locks) and it worked
> fine.
>
> Regards,
> Ovidiu Sas
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
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Re: [SR-Users] Kamailio stability/timing problem w.r.t. registrations?

[Ovidiu Sas]

4.0 (current trunk) is in code freeze.  I would suggest to test the
trunk version (next 4.0).
Even openser 1.3 requires patches to be properly cross compiled.

Regards,
Ovidiu Sas

-- 
VoIP Embedded, Inc.
http://www.voipembedded.com

-- Forwarded message --
From: Sotas Development 
Date: Mon, Jan 28, 2013 at 10:08 AM
Subject: Re: [SR-Users] Kamailio stability/timing problem w.r.t. registrations?
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
Users Mailing List" 


Hi Ovidiu,

Thanks for the warning! We did not yet have much success running the
current master branch, though this may well be a resource problem on
the target platform.

For the moment, we decided to switch back to openser 1.3.5 and wait
for the official 4.0 release.

Regards,
Michiel Veldkamp


On Thu, Jan 17, 2013 at 7:01 PM, Ovidiu Sas  wrote:
>
> If you are running the stable version, there's need for heavy Makefile
> patching in order to properly cross compile (not to include and link
> to host libs).
> The trunk has everything fixed and it's cross-compiling properly for
> most of the modules.
> Make sure that your binaries are properly cross compiled.
>
> Depending on your ARM CPU, atomic locks may or may not work.
> I tested openser without atomic locks (using regular locks) and it worked 
> fine.
>
> Regards,
> Ovidiu Sas
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com

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[SR-Users] Need help understanding/separating signaling from media

[Grant Bagdasarian]

Hello,

I want to separate signaling and media in our architecture but I'm not entirely 
sure if the following will work (note that it is still in a design phase and 
these kinds of constructs are still new to me):

User Agent 1-- Kamailio -- User Agent 2
 |  
|
 |  
|
 |  
|
  -- Media Server Cluster --

The signaling will go through Kamailio but the media stream should pass through 
one of the media servers in the cluster for each call.
Is it possible to achieve this by having Kamailio instruct the media server 
which ports to open and alter sdp information?
I think that the RTP Proxy module does precisely this, but what I still don't 
understand is how the rtp stream is passed between these two agents for each 
call.
Also, is RTP proxy the best way to do this or are there any other third party 
media servers which I could use? I don't need any fancy functionality, just 
basic media handling/pass-through.
Regards,
Grant
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Re: [SR-Users] Error searching $keepalive.socket pvar in NAT Traversal module

[Daniel-Constantin Mierla]

Hello,

what version of kamailio are you using? Do you have '#!KAMAILIO' as 
first line?


You can try the alternative format for the pv, putting everything in 
another group of parenthesis after $, like:


$fs = $(keepalive.socket($du));

Cheers,
Daniel

On 1/28/13 3:31 PM, vito wrote:

Hello,

I started to use Kamailio few months ago. Maybe this is a stupid issue 
but I can´t start Kamailio with NAT TRAVERSAL module and 
$keepalive.socket pseudo-variable.
My script configuration is the default template adding this lines 
(extracted from nat traversal doc):


...
loadmodule "nat_traversal.so"
...

...
$fs = $keepalive.socket($du);
...

I have this error:

-e
 0(17455) ERROR:  [pvapi.c:501]: error searching pvar "keepalive"
 0(17455) ERROR:  [pvapi.c:705]: wrong char [e/101] in 
[$keepalive] at [9 (0)]
 0(17455) :  [cfg.y:3330]: parse error in config file 
/etc/kamailio/kamailio.cfg, line 410, column 8-17: unknown script 
pseudo variable $keepalive

ERROR: bad config file (1 errors)


I tried in this environments:

 Ubuntu 10.04.4 LTS x86_64   Kamailio 3.3.3+git20130128+lucid1
 Ubuntu 10.04.4 LTS i686 Kamailio 3.0.4+lucid1
 Ubuntu 12.04.1 LTS x86_64   Kamailio 3.3.2+precise1
All packages instaled from deb repositories (http://deb.kamailio.org/)

I think there are problem parsing "$keepalive.socket" pseudovariable, 
because Kamailio doesn´t scape this character "." and it doesn´t 
recognise the pseudovar.


Does anyone have this problem?


Regards


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--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, April 16-17, 2013, Berlin
 - http://conference.kamailio.com -

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Re: [SR-Users] Need help understanding/separating signaling from media

[Alex Balashov]

Hi Grant,

On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:


I think that the RTP Proxy module does precisely this, but what I
still don’t understand is how the rtp stream is passed between these
two agents for each call.


The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and 
asks it to engage the call, saying: "Can you please allocate and provide 
me with a pair of ports for each end of this call?"  The rtpproxy 
process answers, and Kamailio then writes the IP and ports provided by 
rtpproxy into the SDP of both the requests and replies involved in 
setting up the call.


The result is that as far as both UAs are concerned, rtpproxy is their 
endpoint, and they will send media to it.  Kamailio instructs rtpproxy 
to bridge both streams.


When the call is torn down, Kamailio tells rtpproxy to disengage and 
deallocate the bridge mapping for those streams.


-- Alex

[1] Which may be running on the same host, or may be running on another 
host, in a distributed fashion.  Kamailio talks to rtpproxy through 
rtpproxy's UDP control socket, and more than one rtpproxy may be used, 
both for failover and round-robin load distribution.  This is readily 
baked into the rtpproxy module.


--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [SR-Users] Need help understanding/separating signaling from media

[Grant Bagdasarian]

Hello Alex,

Thank you for the explanation. It's clear to me now. 
One more thing, I've read that it's not recommended to have media being handled 
by virtual servers, is this also the case when using rtpproxy? We are using 
VMWare for virtualization.

Regards,

Grant


Van: sr-users-boun...@lists.sip-router.org 
[sr-users-boun...@lists.sip-router.org] namens Alex Balashov 
[abalas...@evaristesys.com]
Verzonden: maandag 28 januari 2013 17:11
Aan: sr-users@lists.sip-router.org
Onderwerp: Re: [SR-Users] Need help understanding/separating signaling from 
media

Hi Grant,

On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:

> I think that the RTP Proxy module does precisely this, but what I
> still don’t understand is how the rtp stream is passed between these
> two agents for each call.

The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and
asks it to engage the call, saying: "Can you please allocate and provide
me with a pair of ports for each end of this call?"  The rtpproxy
process answers, and Kamailio then writes the IP and ports provided by
rtpproxy into the SDP of both the requests and replies involved in
setting up the call.

The result is that as far as both UAs are concerned, rtpproxy is their
endpoint, and they will send media to it.  Kamailio instructs rtpproxy
to bridge both streams.

When the call is torn down, Kamailio tells rtpproxy to disengage and
deallocate the bridge mapping for those streams.

-- Alex

[1] Which may be running on the same host, or may be running on another
host, in a distributed fashion.  Kamailio talks to rtpproxy through
rtpproxy's UDP control socket, and more than one rtpproxy may be used,
both for failover and round-robin load distribution.  This is readily
baked into the rtpproxy module.

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [SR-Users] Need help understanding/separating signaling from media

[Alex Balashov]

It is indeed not generally recommended, due to the extremely real-time, 
delay-sensitive nature of media and the way VMs can subject a user space 
process like rtpproxy to distorted I/O scheduling and timing.

However, this seems to be less of a problem as virtualisation evolves closer 
and closer "to the metal", and in any case is unlikely to be an issue for 
relatively small numbers of concurrent calls, on a non-oversubscribed 
hypervisor. 

Still, it is not recommended from a "best practical" point of view, definitely 
not. 

Grant Bagdasarian  wrote:

>Hello Alex,
>
>Thank you for the explanation. It's clear to me now. 
>One more thing, I've read that it's not recommended to have media being
>handled by virtual servers, is this also the case when using rtpproxy?
>We are using VMWare for virtualization.
>
>Regards,
>
>Grant
>
>
>Van: sr-users-boun...@lists.sip-router.org
>[sr-users-boun...@lists.sip-router.org] namens Alex Balashov
>[abalas...@evaristesys.com]
>Verzonden: maandag 28 januari 2013 17:11
>Aan: sr-users@lists.sip-router.org
>Onderwerp: Re: [SR-Users] Need help understanding/separating signaling
>from media
>
>Hi Grant,
>
>On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
>
>> I think that the RTP Proxy module does precisely this, but what I
>> still don’t understand is how the rtp stream is passed between these
>> two agents for each call.
>
>The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and
>asks it to engage the call, saying: "Can you please allocate and
>provide
>me with a pair of ports for each end of this call?"  The rtpproxy
>process answers, and Kamailio then writes the IP and ports provided by
>rtpproxy into the SDP of both the requests and replies involved in
>setting up the call.
>
>The result is that as far as both UAs are concerned, rtpproxy is their
>endpoint, and they will send media to it.  Kamailio instructs rtpproxy
>to bridge both streams.
>
>When the call is torn down, Kamailio tells rtpproxy to disengage and
>deallocate the bridge mapping for those streams.
>
>-- Alex
>
>[1] Which may be running on the same host, or may be running on another
>host, in a distributed fashion.  Kamailio talks to rtpproxy through
>rtpproxy's UDP control socket, and more than one rtpproxy may be used,
>both for failover and round-robin load distribution.  This is readily
>baked into the rtpproxy module.
>
>--
>Alex Balashov - Principal
>Evariste Systems LLC
>235 E Ponce de Leon Ave
>Suite 106
>Decatur, GA 30030
>United States
>Tel: +1-678-954-0670
>Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
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-- 
Sent from my mobile, and thus lacking in the refinement one might expect from a 
fully-fledged keyboard. 

Alex Balashov - Principal 
Evariste Systems LLC 
235 E Ponce de Leon Ave 
Suite 106 
Decatur, GA 30030 
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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[SR-Users] Default 3.3 config and dispatcher

[Paul Belanger]

Greetings List,

First post of many.  I've managed to get up a running a very simple
VoIP network with Kamailio as the core, and asterisk on the edge.
Most of this has been accomplished using the default kamailio.cfg file
shipped in 3.3.

For my next adventure, I'd like to start using the dispatch module to
route calls from Kamailio to asterisk,  however having some trouble
getting my syntax correct.  I was hoping somebody could point me to an
existing post / URL describing what I was asking, if it is based on
the default configuration file, the better.

I should note, if I have Asterisk register to kamailio, I can route
calls fine however, I believe the dispatcher is the better way to do
this.

-- 
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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[SR-Users] Web softphone using websockets against Kamailio

[Pirjo Ahvenainen]

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

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Re: [SR-Users] Need help understanding/separating signaling from media

[Grant Bagdasarian]

Alright, thanks! I'll just use virtuals for our sip proxies and physical 
servers for media, like I intended to do.

Van: sr-users-boun...@lists.sip-router.org 
[sr-users-boun...@lists.sip-router.org] namens Alex Balashov 
[abalas...@evaristesys.com]
Verzonden: maandag 28 januari 2013 18:18
Aan: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users   
Mailing List
Onderwerp: Re: [SR-Users] Need help understanding/separating signaling from 
media

It is indeed not generally recommended, due to the extremely real-time, 
delay-sensitive nature of media and the way VMs can subject a user space 
process like rtpproxy to distorted I/O scheduling and timing.

However, this seems to be less of a problem as virtualisation evolves closer 
and closer "to the metal", and in any case is unlikely to be an issue for 
relatively small numbers of concurrent calls, on a non-oversubscribed 
hypervisor.

Still, it is not recommended from a "best practical" point of view, definitely 
not.

Grant Bagdasarian  wrote:

>Hello Alex,
>
>Thank you for the explanation. It's clear to me now.
>One more thing, I've read that it's not recommended to have media being
>handled by virtual servers, is this also the case when using rtpproxy?
>We are using VMWare for virtualization.
>
>Regards,
>
>Grant
>
>
>Van: sr-users-boun...@lists.sip-router.org
>[sr-users-boun...@lists.sip-router.org] namens Alex Balashov
>[abalas...@evaristesys.com]
>Verzonden: maandag 28 januari 2013 17:11
>Aan: sr-users@lists.sip-router.org
>Onderwerp: Re: [SR-Users] Need help understanding/separating signaling
>from media
>
>Hi Grant,
>
>On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
>
>> I think that the RTP Proxy module does precisely this, but what I
>> still don’t understand is how the rtp stream is passed between these
>> two agents for each call.
>
>The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and
>asks it to engage the call, saying: "Can you please allocate and
>provide
>me with a pair of ports for each end of this call?"  The rtpproxy
>process answers, and Kamailio then writes the IP and ports provided by
>rtpproxy into the SDP of both the requests and replies involved in
>setting up the call.
>
>The result is that as far as both UAs are concerned, rtpproxy is their
>endpoint, and they will send media to it.  Kamailio instructs rtpproxy
>to bridge both streams.
>
>When the call is torn down, Kamailio tells rtpproxy to disengage and
>deallocate the bridge mapping for those streams.
>
>-- Alex
>
>[1] Which may be running on the same host, or may be running on another
>host, in a distributed fashion.  Kamailio talks to rtpproxy through
>rtpproxy's UDP control socket, and more than one rtpproxy may be used,
>both for failover and round-robin load distribution.  This is readily
>baked into the rtpproxy module.
>
>--
>Alex Balashov - Principal
>Evariste Systems LLC
>235 E Ponce de Leon Ave
>Suite 106
>Decatur, GA 30030
>United States
>Tel: +1-678-954-0670
>Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
>
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--
Sent from my mobile, and thus lacking in the refinement one might expect from a 
fully-fledged keyboard.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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[SR-Users] REGISTER OK but on send "407 Proxy Authentication Required" with no ACK

[Carsten Maass]

Hi all,

I am trying to set up a fax gateway in the following way:

PSTN-GW (10.1.1.150) <--> Kamailio (10.1.1.148:5123) <--> t38modem
(10.1.1.148:6050) <--> Hylafax

PSTN-GW is a standalone Berofix appliance and Kamailio 3.3.3, t38modem
1.2 and Hylafax 6.0.3 running on the same host under Redhat EL6.

I used the default kamailio.cfg and just adjusted the PSTN route pattern
to "if(!($rU=~"^(\+|00|0)[1-9][0-9]{3,20}$"))", to route all calls
starting with 0 to the PSTN-GW.

Both PSTN-GW and t38modem Successfully register to Kamailio but when I
try to send out a fax from t38modem, Kamailio doesn't ACK the
Proxy-Authorization request and t38modem terminates the call:



05:49:11.158157 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto
UDP (17), length 986)
10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 958
E.@.@.Y.
.d.
.d.INVITE sip:030987654321@10.1.1.148:5123 SIP/2.0
Route: 
Date: Tue, 29 Jan 2013 04:49:11 GMT
CSeq: 1 INVITE
Via: SIP/2.0/UDP
10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport
User-Agent: T38Modem/1.2.0
From: "root"
;tag=169d43e4-3c68-e211-98e4-005056bd002b
Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b@myhost.mydomain.local
Organization: Vyacheslav Frolov
To: 
Contact: 
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 262
Max-Forwards: 70

v=0
o=- 1359434951 1 IN IP4 10.1.1.148
s=Opal SIP Session
c=IN IP4 10.1.1.148
t=0 0
m=audio 5008 RTP/AVP 8 101 100
a=sendrecv
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193

05:49:11.158927 IP (tos 0x10, ttl 64, id 28825, offset 0, flags [none],
proto UDP (17), length 584)
10.1.1.148.5123 > 10.1.1.148.6050: UDP, length 556
E..Hp...@.*.
.d.
.d..4.oSIP/2.0 407 Proxy Authentication Required
CSeq: 1 INVITE
Via: SIP/2.0/UDP
10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport=6050
From: "root"
;tag=169d43e4-3c68-e211-98e4-005056bd002b
Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b@myhost.mydomain.local
To:
;tag=75c88a497c1ef184e8ac0d2e60c130e4.0580
Proxy-Authenticate: Digest realm="10.1.1.148",
nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W"
Server: kamailio (3.3.3 (x86_64/linux))
Content-Length: 0


05:49:11.160273 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto
UDP (17), length 501)
10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 473
E.@.@.[.
.d.
.d.ACK sip:030987654321@10.1.1.148:5123 SIP/2.0
Route: 
CSeq: 1 ACK
Via: SIP/2.0/UDP
10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport
From: "root"
;tag=169d43e4-3c68-e211-98e4-005056bd002b
Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b@myhost.mydomain.local
To:
;tag=75c88a497c1ef184e8ac0d2e60c130e4.0580
Content-Length: 0
Max-Forwards: 70


05:49:11.166542 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto
UDP (17), length 1195)
10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 1167
E.@.@.Y.
.d.
.d.INVITE sip:030987654321@10.1.1.148:5123 SIP/2.0
Route: 
Date: Tue, 29 Jan 2013 04:49:11 GMT
CSeq: 2 INVITE
Via: SIP/2.0/UDP
10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport
User-Agent: T38Modem/1.2.0
From: "root"
;tag=169d43e4-3c68-e211-98e4-005056bd002b
Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b@myhost.mydomain.local
Organization: Vyacheslav Frolov
To: 
Contact: 
Proxy-Authorization: Digest username="979", realm="10.1.1.148",
nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W",
uri="sip:030987654321@10.1.1.148:5123", algorithm=MD5,
response="48d896ef41a71720b29c56230858582f"
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 262
Max-Forwards: 70

v=0
o=- 1359434951 1 IN IP4 10.1.1.148
s=Opal SIP Session
c=IN IP4 10.1.1.148
t=0 0
m=audio 5008 RTP/AVP 8 101 100
a=sendrecv
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193

05:49:11.167573 IP (tos 0x10, ttl 64, id 28826, offset 0, flags [none],
proto UDP (17), length 584)
10.1.1.148.5123 > 10.1.1.148.6050: UDP, length 556
E..Hp...@.*.
.d.
.d..4.oSIP/2.0 407 Proxy Authentication Required
CSeq: 2 INVITE
Via: SIP/2.0/UDP
10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport=6050
From: "root"
;tag=169d43e4-3c68-e211-98e4-005056bd002b
Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b@myhost.mydomain.local
To:
;tag=75c88a497c1ef184e8ac0d2e60c130e4.263a
Proxy-Authenticate: Digest realm="10.1.1.148",
nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W"
Server: kamailio (3.3.3 (x86_64/linux))
Content-Length: 0


05:49:11.169220 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto
UDP (17), length 710)
10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 682
E.@.@.Z.
.d.
.d.ACK sip:030987654321@10.1.1.148:5123 SIP/2.0
Route: 
CSeq: 2 ACK
Via: SIP/2.0/UDP
10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport
From: "root"
;ta

Re: [SR-Users] Major problem with setup NAT + MULTIDOMAIN

[SamyGo]

Hi Rumen,

Can you tell how are you creating users in your "sip" table? Are you sure
that the passwords are calculated using the real/domain part of the SIP
User definition?

Also ensure that your auth_db module has multidomain(use-domain) modparam
enabled and if you're using your own table coloumn to store user's domain
part then edit the modparam to use that particular coloumn (domain_coloumn)

Thanks,
Sammy



On Tue, Jan 29, 2013 at 2:52 AM, Rumen Mihailov wrote:

> Hi guys,
>
> I've baning my head for the last 6 7 days with something that have to
> be universally simple but obviously for me it is not, so please do
> help.
>
> I have the following setup.
>
> xx.xx.xx.xx (real IP 1:1 NAT) -> 10.2.47.201
>
> Kamailio listens on 10.2.47.201
> I have setup a domain xxx.itradebg.com and I've made the appropriate
> records in the DNS.
> I have inserted the domain xxx.itradebg.com in the domain table in the
> database and kamctl domain show shows the correct thing.
>
> Now when I try to register 9...@xxx.itradebg.com I get Request timed out
>
> I have this in my config:
> if (is_method("REGISTER") || from_uri==myself)
> {
> if (!auth_check("$fd", "sip", "1")) {
> auth_challenge("$fd", "0");
> exit;
> }
> }
> if (from_uri!=myself && uri!=myself)
> {
>
> if(lookup_domain("$fd", "@from.uri.host")){
> xlog("LOCAL DOMAIN -> REGISTER");
> exit;
> }
> else{
> xlog("L_WARN","FROM NOT LOCAL");
> sl_send_reply("403","Sorry mate not relaying");
> exit;
> }
> }
>
> By the time of writing this I realised that this should be all
> rewritten to this only, as the second check is useless:
>
> if (is_method("REGISTER") || from_uri==myself)
> {
> if (!auth_check("$fd", "sip", "1")) {
> auth_challenge("$fd", "0");
> exit;
> }
> }
>
> However I get 401 Unauthorized. The previous setup was giving me 408 -
> Request Timed out...
>
> I am pretty sure I have the passwords correct.
>
> What I cannot understand is why authentication fails ?!
>
> Also, how do I set kamailio to print debug output ?
> debug=9
> log_stderror=no
>
> This is what I have in the .conf file. /var/log/messages seems to only
> have what I have printed with xlog... (think)
>
> I am really out of ideas... perhaps I need some sleep.
>
> Anyway any input is helpful guys.
>
> Thanks,
> Rumen
> http://itradebg.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
>
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[SR-Users] Sip express live Cd router

[Shoukat, Kashif]

Dear ,

Can you please provide me link of sip express live Cd router .. I want to use 
it... waiting your kind response

Regards,
Kashif Shoukat | Analyst Software Engineer |
Work :   + 1.202.289.9898 x 9077
Mobile :+ 92.323.431.8548
Skype:   kashif.shoukat.dgs
www.dgsworld.com

dgs
 deliberate by design

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