date:20141218

[SR-Users] Kamailio CDRs

2014-12-18 Thread Massimo Varriale (IPZeta)

Hi Guys!
I'm trying to store CDRs for Kamailio calls.

Following Siremis tutorial (http://siremis.asipto.com/install-accounting/)  I'm 
able to gather Succesfull Calls and using the MySQL stored procedure I can have 
a CDR table. 
But what I can see now is that only outbound calls with a duration are stored 
and that failed attempts are not stored into the database.
How to capture those attempts too?

Here is my accounting configuration:

# - acc params -
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
modparam("acc", "detect_direction", 0)
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra", 
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
"src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd;src_ip=$si")
modparam("acc", "cdr_enable", 1)
#!endif


Thanks
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Re: [SR-Users] Kamailio CDRs

2014-12-18 Thread Marino Mileti

Hi Massimo

you've to modify also your scripts and set flag FLT_ACCMISSED when you want
to log the missed or failed call

Marino



-
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Re: [SR-Users] SIP Fragments

2014-12-18 Thread Olle E. Johansson


On 18 Dec 2014, at 03:01, Alex Balashov  wrote:

> Indeed, gzcompress is a Kamailio-only concept. 
In fact we got it from Facetime...

SHould be easy to implement in Asterisk too.
/O
> 
> On 17 December 2014 20:58:22 GMT-05:00, Marc Soda  wrote:
> So gzcompress is no good with Asterisk then?  Is that meant to be used only 
> with another Kamailio proxy?
> 
> We're trying to do a WebRTC POC with Kamailio as the proxy.  The SIP headers 
> and SDP are huge!  I've never seen such big messages.
> 
> Thanks,
> Marc
> 
> On Wed, Dec 17, 2014 at 6:47 PM, Daniel-Constantin Mierla  
> wrote:
> On 17/12/14 23:20, Alex Balashov wrote:
> > On 12/17/2014 05:14 PM, Marc Soda wrote:
> >
> >> I'm having a problem reassembling UDP packets on my Asterisk servers
> >> after passing through Kamailio (it appears to me an OS level issue,
> >> nothing to do with Kamailio).  Is there a way, with Kamailio, to limit
> >> the size of a SIP message?  I know I can just start removing headers,
> >> but that doesn't seem like a realistic solution.  I see that Kamailio
> >> can compress the message body, but can Asterisk handle that?  How do
> >> other people handle this?
> >
> > 1. Any SIP-compliant endpoint should be able to handle compact
> > headers. Per RFC 3261 7.3.3 ("Compact Form"):
> >
> >Implementations MUST accept both the long and short forms of
> >each header name.
> 
> I don't think compact names for headers or joining bodies under single
> header name helps that much, it would be in the range of few tens of bytes.
> 
> >
> > 2. Some headers are critical should not be removed. Others really are
> > mostly useless bloat commonly added by verbose UACs, and, practically
> > speaking, the other peer will be neither colder nor warmer if they are
> > removed, unless there is a specific use for them.
> >
> > Good candidates are:
> >
> > a) The "Date" header.
> > b) Accept: headers listing every MIME type in the known universe.
> 
> Mentioned on my previous email too -- keep_hf() from textopsx module can
> be handy here.
> 
> >
> > 3. If one or more of your endpoints offer every codec in the known
> > universe in the SDP, you can restrict the codecs offered to reduce the
> > SDP size.
> 
> Another option to reduce the size -- sdpops module has related functions
> for sdp management.
> 
> >
> > 4. You could use TCP. In fact, RFC 3261 actually mandates this. Per
> > RFC 3261 Section 18.1.1 ("Sending Requests"):
> >
> >If a request is within 200 bytes of the path MTU, or if it is larger
> >than 1300 bytes and the path MTU is unknown, the request MUST be sent
> >using an RFC 2914 [43] congestion controlled transport protocol, such
> >as TCP.
> >
> > Of course, in reality, nobody cares or follows this, and many SIP
> > endpoints don't even support TCP (also mandated by RFC 3261).
> >
> > 5. In some situations, header bloat comes from requests passing
> > through numerous proxies, each of which add a stackable Via header
> > and, if applicable, a Route/Record-Route set.
> >
> > Reducing the number of intermediate proxies can help with this.
> >
> > 6. You could run the traffic through a lightweight, signalling-only
> > B2BUA, such as SEMS, which deals with fragmented UDP in incoming
> > requests just fine, but does not reoriginate on leg B all the bloated
> > headers that came in on leg A.
> 
> SEMS (like any other application layer program) had very few to do with
> fragmentation. It is the kernel/operating system that sorts all this. It
> the application is the same 'recvfrom(...)'.
> 
> At the end, Asterisk is also a B2BUA and I guess if there is a server
> with an OS that can handle udp fragmentation, the Asterisk will be run
> there instead of adding another b2bua.
> 
> Cheers,
> Daniel
> 
> >
> > 7. Other than these things, there are no real solutions.
> >
> > -- Alex
> >
> 
> 
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> 
> 
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> 
> 
> 
> 
> 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
> 
> --
> 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: +1-678-954-0671
> Web: http://www.evaristesys.com/, http://www.alexbalashov.com
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Re: [SR-Users] SIP Fragments

2014-12-18 Thread Daniel-Constantin Mierla


On 18/12/14 02:58, Marc Soda wrote:
> So gzcompress is no good with Asterisk then?  Is that meant to be used
> only with another Kamailio proxy?

Apparently Apple Facetime is using this kind of compression (as it was
reported on a blog and triggered the implementation in Kamailio), but
one cannot interconnect with them anyhow. IIRC, FreeSwitch implemented
it as well.

>
> We're trying to do a WebRTC POC with Kamailio as the proxy.  The SIP
> headers and SDP are huge!  I've never seen such big messages.

This is the web world -- lot of data even for little content, like for
html pages :-)

Cheers,
Daniel

>
> Thanks,
> Marc
>
> On Wed, Dec 17, 2014 at 6:47 PM, Daniel-Constantin Mierla
> mailto:mico...@gmail.com>> wrote:
>
> On 17/12/14 23:20, Alex Balashov wrote:
> > On 12/17/2014 05:14 PM, Marc Soda wrote:
> >
> >> I'm having a problem reassembling UDP packets on my Asterisk
> servers
> >> after passing through Kamailio (it appears to me an OS level issue,
> >> nothing to do with Kamailio).  Is there a way, with Kamailio,
> to limit
> >> the size of a SIP message?  I know I can just start removing
> headers,
> >> but that doesn't seem like a realistic solution.  I see that
> Kamailio
> >> can compress the message body, but can Asterisk handle that? 
> How do
> >> other people handle this?
> >
> > 1. Any SIP-compliant endpoint should be able to handle compact
> > headers. Per RFC 3261 7.3.3 ("Compact Form"):
> >
> >Implementations MUST accept both the long and short forms of
> >each header name.
>
> I don't think compact names for headers or joining bodies under single
> header name helps that much, it would be in the range of few tens
> of bytes.
>
> >
> > 2. Some headers are critical should not be removed. Others
> really are
> > mostly useless bloat commonly added by verbose UACs, and,
> practically
> > speaking, the other peer will be neither colder nor warmer if
> they are
> > removed, unless there is a specific use for them.
> >
> > Good candidates are:
> >
> > a) The "Date" header.
> > b) Accept: headers listing every MIME type in the known universe.
>
> Mentioned on my previous email too -- keep_hf() from textopsx
> module can
> be handy here.
>
> >
> > 3. If one or more of your endpoints offer every codec in the known
> > universe in the SDP, you can restrict the codecs offered to
> reduce the
> > SDP size.
>
> Another option to reduce the size -- sdpops module has related
> functions
> for sdp management.
>
> >
> > 4. You could use TCP. In fact, RFC 3261 actually mandates this. Per
> > RFC 3261 Section 18.1.1 ("Sending Requests"):
> >
> >If a request is within 200 bytes of the path MTU, or if it is
> larger
> >than 1300 bytes and the path MTU is unknown, the request MUST
> be sent
> >using an RFC 2914 [43] congestion controlled transport
> protocol, such
> >as TCP.
> >
> > Of course, in reality, nobody cares or follows this, and many SIP
> > endpoints don't even support TCP (also mandated by RFC 3261).
> >
> > 5. In some situations, header bloat comes from requests passing
> > through numerous proxies, each of which add a stackable Via header
> > and, if applicable, a Route/Record-Route set.
> >
> > Reducing the number of intermediate proxies can help with this.
> >
> > 6. You could run the traffic through a lightweight, signalling-only
> > B2BUA, such as SEMS, which deals with fragmented UDP in incoming
> > requests just fine, but does not reoriginate on leg B all the
> bloated
> > headers that came in on leg A.
>
> SEMS (like any other application layer program) had very few to do
> with
> fragmentation. It is the kernel/operating system that sorts all
> this. It
> the application is the same 'recvfrom(...)'.
>
> At the end, Asterisk is also a B2BUA and I guess if there is a server
> with an OS that can handle udp fragmentation, the Asterisk will be run
> there instead of adding another b2bua.
>
> Cheers,
> Daniel
>
> >
> > 7. Other than these things, there are no real solutions.
> >
> > -- Alex
> >
>
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda  -
> http://www.linkedin.com/in/miconda
>
>
> ___
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> list
> sr-users@lists.sip-router.org 
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Re: [SR-Users] Kamailio CDRs

2014-12-18 Thread Massimo Varriale (IPZeta)

Hi Marino,
now  the parameter is set to number 2

#!define FLT_ACCMISSED 2

Should raised to 3 or more?

Thanks




Il giorno 18/dic/2014, alle ore 11:02, Marino Mileti ha scritto:

> Hi Massimo
> 
> you've to modify also your scripts and set flag FLT_ACCMISSED when you want
> to log the missed or failed call
> 
> Marino
> 
> 
> 
> -
> Marino Mileti
> --
> View this message in context: 
> http://sip-router.1086192.n5.nabble.com/Kamailio-CDRs-tp11p12.html
> Sent from the Users mailing list archive at Nabble.com.
> 
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Re: [SR-Users] Kamailio CDRs

2014-12-18 Thread Marino Mileti

No no you've to edit you scripts...
For examplein a failure_route:

if (t_check_status("408")) {
#!ifdef WITH_ACCOUNTING   
setflag(FLT_ACCFAILED);
#!endif
}

In this way you can "log" a call missed for timeout.


Massimo Varriale (IPZeta) wrote
> Hi Marino,
> now  the parameter is set to number 2
> 
> #!define FLT_ACCMISSED 2
> 
> Should raised to 3 or more?
> 
> Thanks





-
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--
View this message in context: 
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Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

2014-12-18 Thread José Seabra

Hello Daniel,
Do you need more information from my kamailio settup, in order to try
understand why this issue happens?
Thank you
BR
José Seabra


2014-12-16 12:16 GMT+00:00 José Seabra :
>
> Update:
> My last email has the wrong  example, the correct code is:
>
> if(!ds_select_dst("9", "4"))
> {
> xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
> sl_send_reply("503", "No proxy available, try again later");
> exit;
> }
>
> I'm using algorithm 4.
>
> 2014-12-16 12:14 GMT+00:00 José Seabra :
>>
>> Hello Daniel,
>>
>> Yes it enter inside a condition.
>>
>> my code is:
>>
>> if(!ds_select_dst("9", "0"))
>> {
>> xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
>> sl_send_reply("503", "No proxy available, try again later");
>> exit;
>> }
>> and all sip registrers sent receives a sip reply 503 "No proxy
>> available, try again later"
>>
>>
>> BR
>> José Seabra
>>
>> 2014-12-16 12:06 GMT+00:00 Daniel-Constantin Mierla :
>>>
>>>  Hello,
>>>
>>> what do you mean starts returning null?
>>>
>>> Does it enter inside a condition like:
>>>
>>> if(!ds_select_dst(...))
>>>
>>> Are there any error messages in syslog?
>>>
>>> Daniel
>>>
>>>
>>> On 16/12/14 12:51, José Seabra wrote:
>>>
>>> Hello there,
>>>
>>>  I'm doing stress testing against kamailio (sip load balancer role)
>>> sending sip registrar, my stress test software is sending 50 registrations
>>> per second, after 30 seconds and 1000 registers processed the function
>>> ds_select_dst (I'm using algorithm 4) returns null for all registers sent
>>> from my software, and all registrar sip servers are available and in active
>>> mode, if I reduce the software to  send 20 registrar's per second, the
>>> issue only happens after 60 seconds and kamailio (sip load balancer role)
>>>  process 4000 registrers, after this, function ds_select_dst starts
>>> returning null
>>>
>>>
>>>  Anyone has any idea why it is happening?
>>> My kamailio version is 4.2
>>>
>>>  Best Regards
>>>  --
>>>  Cumprimentos
>>> José Seabra
>>>
>>>
>>> ___
>>> 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 Mierlahttp://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
>>>
>>>
>>
>> --
>> Cumprimentos
>> José Seabra
>>
>
>
> --
> Cumprimentos
> José Seabra
>


-- 
Cumprimentos
José Seabra
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Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

2014-12-18 Thread Daniel-Constantin Mierla

Hello,

I asked in the first email - do you get error messages in syslog? If
not, then you have to run with debug=3 and look at the log messages to
see if you get further hints.

Cheers,
Daniel

On 18/12/14 12:29, José Seabra wrote:
> Hello Daniel,
> Do you need more information from my kamailio settup, in order to try
> understand why this issue happens?
> Thank you
> BR
> José Seabra
>
>
> 2014-12-16 12:16 GMT+00:00 José Seabra  >:
>
> Update:
> My last email has the wrong  example, the correct code is:
>
> if(!ds_select_dst("9", "4"))
> {
> xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
> sl_send_reply("503", "No proxy available, try again later");
> exit;
> }
>
> I'm using algorithm 4.
>
> 2014-12-16 12:14 GMT+00:00 José Seabra  >:
>
> Hello Daniel,
>
> Yes it enter inside a condition.
>
> my code is:
>
> if(!ds_select_dst("9", "0"))
> {
> xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
> sl_send_reply("503", "No proxy available, try again later");
> exit;
> }
> and all sip registrers sent receives a sip reply 503 "No proxy
> available, try again later"
>
>
> BR
> José Seabra
>
> 2014-12-16 12:06 GMT+00:00 Daniel-Constantin Mierla
> mailto:mico...@gmail.com>>:
>
> Hello,
>
> what do you mean starts returning null?
>
> Does it enter inside a condition like:
>
> if(!ds_select_dst(...))
>
> Are there any error messages in syslog?
>
> Daniel
>
>
> On 16/12/14 12:51, José Seabra wrote:
>> Hello there,
>>
>> I'm doing stress testing against kamailio (sip load
>> balancer role) sending sip registrar, my stress test
>> software is sending 50 registrations per second, after 30
>> seconds and 1000 registers processed the function
>> ds_select_dst (I'm using algorithm 4) returns null for
>> all registers sent from my software, and all registrar
>> sip servers are available and in active mode, if I reduce
>> the software to  send 20 registrar's per second, the
>> issue only happens after 60 seconds and kamailio (sip
>> load balancer role)  process 4000 registrers, after this,
>> function ds_select_dst starts returning null
>>
>>
>> Anyone has any idea why it is happening?
>> My kamailio version is 4.2
>>
>> Best Regards
>> -- 
>> Cumprimentos
>> José Seabra
>>
>>
>> ___
>> 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://twitter.com/#!/miconda  - 
> http://www.linkedin.com/in/miconda
>
>
> ___
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> mailing list
> sr-users@lists.sip-router.org
> 
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>
>
>
> -- 
> Cumprimentos
> José Seabra
>
>
>
> -- 
> Cumprimentos
> José Seabra
>
>
>
> -- 
> Cumprimentos
> José Seabra

-- 
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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[SR-Users] Kamailio World 2015 – Call For Speakers

2014-12-18 Thread Daniel-Constantin Mierla

Hello,

a short note to announce for everyone interested that submissions of
presentation proposals is now open for Kamailio World 2015 (Berlin, May
27-29). More details at:

  -
http://conference.kamailio.com/k03/2014/12/kamailio-world-2015-call-for-speakers/

Cheers,
Daniel

-- 
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http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


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Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

2014-12-18 Thread José Seabra

Hello Daniel,
Thank you for your reply,

Please check out the attachment with kamailio logs in mode debug 3, i
didn't find any errors, the only thingh that is confusing me is this line:
/usr/local/sbin/kamailio[13690]: DEBUG: dispatcher [dispatch.c:1745]:
ds_select_dst_limit(): alg hash [0]

I don't know if this "ds_select_dst_limit(): alg hash [0]" is related with
algorithm parameter in function ds_select_dst, but, if yes i cannot
understand why it is 0 because I'm using algorithm 4.

BR
José Seabra


2014-12-18 11:35 GMT+00:00 Daniel-Constantin Mierla :
>
>  Hello,
>
> I asked in the first email - do you get error messages in syslog? If not,
> then you have to run with debug=3 and look at the log messages to see if
> you get further hints.
>
> Cheers,
> Daniel
>
>
> On 18/12/14 12:29, José Seabra wrote:
>
> Hello Daniel,
> Do you need more information from my kamailio settup, in order to try
> understand why this issue happens?
> Thank you
> BR
> José Seabra
>
>
> 2014-12-16 12:16 GMT+00:00 José Seabra :
>>
>> Update:
>> My last email has the wrong  example, the correct code is:
>>
>>  if(!ds_select_dst("9", "4"))
>>  {
>>  xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
>>  sl_send_reply("503", "No proxy available, try again later");
>>  exit;
>> }
>>
>>  I'm using algorithm 4.
>>
>> 2014-12-16 12:14 GMT+00:00 José Seabra :
>>>
>>> Hello Daniel,
>>>
>>> Yes it enter inside a condition.
>>>
>>>  my code is:
>>>
>>>  if(!ds_select_dst("9", "0"))
>>> {
>>>  xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
>>>  sl_send_reply("503", "No proxy available, try again later");
>>>  exit;
>>> }
>>> and all sip registrers sent receives a sip reply 503 "No proxy
>>> available, try again later"
>>>
>>>
>>>  BR
>>> José Seabra
>>>
>>> 2014-12-16 12:06 GMT+00:00 Daniel-Constantin Mierla :
>>>

  Hello,

 what do you mean starts returning null?

 Does it enter inside a condition like:

 if(!ds_select_dst(...))

 Are there any error messages in syslog?

 Daniel


 On 16/12/14 12:51, José Seabra wrote:

  Hello there,

  I'm doing stress testing against kamailio (sip load balancer role)
 sending sip registrar, my stress test software is sending 50 registrations
 per second, after 30 seconds and 1000 registers processed the function
 ds_select_dst (I'm using algorithm 4) returns null for all registers sent
 from my software, and all registrar sip servers are available and in active
 mode, if I reduce the software to  send 20 registrar's per second, the
 issue only happens after 60 seconds and kamailio (sip load balancer role)
  process 4000 registrers, after this, function ds_select_dst starts
 returning null


  Anyone has any idea why it is happening?
 My kamailio version is 4.2

  Best Regards
  --
  Cumprimentos
 José Seabra


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 listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


 --
 Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
 http://www.linkedin.com/in/miconda


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>>>
>>>  --
>>>  Cumprimentos
>>> José Seabra
>>>
>>
>>
>>   --
>> Cumprimentos
>> José Seabra
>>
>
>
>  --
> Cumprimentos
> José Seabra
>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
>
>

-- 
Cumprimentos
José Seabra


dispatcherdebug
Description: Binary data
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Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

2014-12-18 Thread Carsten Bock

Hi,

quick questions:
- Do you have the Dispatcher Activity detection turned on?
- Do you do something like "ds_mark_dst()" in your failure route?
- Can you send me your config, so i can verify a few things?

Thanks,
Carsten



2014-12-18 15:14 GMT+01:00 José Seabra :
> Hello Daniel,
> Thank you for your reply,
>
> Please check out the attachment with kamailio logs in mode debug 3, i didn't
> find any errors, the only thingh that is confusing me is this line:
> /usr/local/sbin/kamailio[13690]: DEBUG: dispatcher [dispatch.c:1745]:
> ds_select_dst_limit(): alg hash [0]
>
> I don't know if this "ds_select_dst_limit(): alg hash [0]" is related with
> algorithm parameter in function ds_select_dst, but, if yes i cannot
> understand why it is 0 because I'm using algorithm 4.
>
> BR
> José Seabra
>
>
> 2014-12-18 11:35 GMT+00:00 Daniel-Constantin Mierla :
>>
>> Hello,
>>
>> I asked in the first email - do you get error messages in syslog? If not,
>> then you have to run with debug=3 and look at the log messages to see if you
>> get further hints.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 18/12/14 12:29, José Seabra wrote:
>>
>> Hello Daniel,
>> Do you need more information from my kamailio settup, in order to try
>> understand why this issue happens?
>> Thank you
>> BR
>> José Seabra
>>
>>
>> 2014-12-16 12:16 GMT+00:00 José Seabra :
>>>
>>> Update:
>>> My last email has the wrong  example, the correct code is:
>>>
>>> if(!ds_select_dst("9", "4"))
>>> {
>>> xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
>>> sl_send_reply("503", "No proxy available, try again later");
>>> exit;
>>> }
>>>
>>> I'm using algorithm 4.
>>>
>>> 2014-12-16 12:14 GMT+00:00 José Seabra :

 Hello Daniel,

 Yes it enter inside a condition.

 my code is:

 if(!ds_select_dst("9", "0"))
 {
 xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
 sl_send_reply("503", "No proxy available, try again later");
 exit;
 }
 and all sip registrers sent receives a sip reply 503 "No proxy
 available, try again later"


 BR
 José Seabra

 2014-12-16 12:06 GMT+00:00 Daniel-Constantin Mierla :
>
> Hello,
>
> what do you mean starts returning null?
>
> Does it enter inside a condition like:
>
> if(!ds_select_dst(...))
>
> Are there any error messages in syslog?
>
> Daniel
>
>
> On 16/12/14 12:51, José Seabra wrote:
>
> Hello there,
>
> I'm doing stress testing against kamailio (sip load balancer role)
> sending sip registrar, my stress test software is sending 50 registrations
> per second, after 30 seconds and 1000 registers processed the function
> ds_select_dst (I'm using algorithm 4) returns null for all registers sent
> from my software, and all registrar sip servers are available and in 
> active
> mode, if I reduce the software to  send 20 registrar's per second, the 
> issue
> only happens after 60 seconds and kamailio (sip load balancer role)  
> process
> 4000 registrers, after this, function ds_select_dst starts returning null
>
>
> Anyone has any idea why it is happening?
> My kamailio version is 4.2
>
> Best Regards
> --
> Cumprimentos
> José Seabra
>
>
> ___
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>
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
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>


 --
 Cumprimentos
 José Seabra
>>>
>>>
>>>
>>> --
>>> Cumprimentos
>>> José Seabra
>>
>>
>>
>> --
>> Cumprimentos
>> José Seabra
>>
>>
>> --
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
>
> --
> Cumprimentos
> José Seabra
>
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Schomburgstr. 80
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Re: [SR-Users] SIP Fragments

2014-12-18 Thread Marc Soda

Content-Length: 1901.

So trimming up the headers isn't going to get me anywhere...  I'm not
comfortable enough with WebRTC to know what to trim out of the SDP, either.

How can I force Kamailio to use TCP for SIP when relaying the call?  I
haven't found much info on it.

Marc

On Thu, Dec 18, 2014 at 5:06 AM, Daniel-Constantin Mierla  wrote:
>
>
> On 18/12/14 02:58, Marc Soda wrote:
>
> So gzcompress is no good with Asterisk then?  Is that meant to be used
> only with another Kamailio proxy?
>
>
> Apparently Apple Facetime is using this kind of compression (as it was
> reported on a blog and triggered the implementation in Kamailio), but one
> cannot interconnect with them anyhow. IIRC, FreeSwitch implemented it as
> well.
>
>
>  We're trying to do a WebRTC POC with Kamailio as the proxy.  The SIP
> headers and SDP are huge!  I've never seen such big messages.
>
>
> This is the web world -- lot of data even for little content, like for
> html pages :-)
>
> Cheers,
> Daniel
>
>
>
>  Thanks,
> Marc
>
> On Wed, Dec 17, 2014 at 6:47 PM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>>
>> On 17/12/14 23:20, Alex Balashov wrote:
>>  > On 12/17/2014 05:14 PM, Marc Soda wrote:
>> >
>> >> I'm having a problem reassembling UDP packets on my Asterisk servers
>> >> after passing through Kamailio (it appears to me an OS level issue,
>> >> nothing to do with Kamailio).  Is there a way, with Kamailio, to limit
>> >> the size of a SIP message?  I know I can just start removing headers,
>> >> but that doesn't seem like a realistic solution.  I see that Kamailio
>> >> can compress the message body, but can Asterisk handle that?  How do
>> >> other people handle this?
>> >
>>  > 1. Any SIP-compliant endpoint should be able to handle compact
>> > headers. Per RFC 3261 7.3.3 ("Compact Form"):
>> >
>> >Implementations MUST accept both the long and short forms of
>> >each header name.
>>
>> I don't think compact names for headers or joining bodies under single
>> header name helps that much, it would be in the range of few tens of
>> bytes.
>>
>> >
>> > 2. Some headers are critical should not be removed. Others really are
>> > mostly useless bloat commonly added by verbose UACs, and, practically
>> > speaking, the other peer will be neither colder nor warmer if they are
>> > removed, unless there is a specific use for them.
>> >
>> > Good candidates are:
>> >
>> > a) The "Date" header.
>> > b) Accept: headers listing every MIME type in the known universe.
>>
>> Mentioned on my previous email too -- keep_hf() from textopsx module can
>> be handy here.
>>
>> >
>> > 3. If one or more of your endpoints offer every codec in the known
>> > universe in the SDP, you can restrict the codecs offered to reduce the
>> > SDP size.
>>
>> Another option to reduce the size -- sdpops module has related functions
>> for sdp management.
>>
>> >
>> > 4. You could use TCP. In fact, RFC 3261 actually mandates this. Per
>> > RFC 3261 Section 18.1.1 ("Sending Requests"):
>> >
>> >If a request is within 200 bytes of the path MTU, or if it is larger
>> >than 1300 bytes and the path MTU is unknown, the request MUST be sent
>> >using an RFC 2914 [43] congestion controlled transport protocol, such
>> >as TCP.
>> >
>> > Of course, in reality, nobody cares or follows this, and many SIP
>> > endpoints don't even support TCP (also mandated by RFC 3261).
>> >
>> > 5. In some situations, header bloat comes from requests passing
>> > through numerous proxies, each of which add a stackable Via header
>> > and, if applicable, a Route/Record-Route set.
>> >
>> > Reducing the number of intermediate proxies can help with this.
>> >
>> > 6. You could run the traffic through a lightweight, signalling-only
>> > B2BUA, such as SEMS, which deals with fragmented UDP in incoming
>> > requests just fine, but does not reoriginate on leg B all the bloated
>> > headers that came in on leg A.
>>
>> SEMS (like any other application layer program) had very few to do with
>> fragmentation. It is the kernel/operating system that sorts all this. It
>> the application is the same 'recvfrom(...)'.
>>
>> At the end, Asterisk is also a B2BUA and I guess if there is a server
>> with an OS that can handle udp fragmentation, the Asterisk will be run
>> there instead of adding another b2bua.
>>
>> Cheers,
>> Daniel
>>
>> >
>> > 7. Other than these things, there are no real solutions.
>> >
>> > -- Alex
>>  >
>>
>>
>> --
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>>
>>
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>>
>
>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] SIP Fragments

2014-12-18 Thread Juha Heinanen

Marc Soda writes:

> How can I force Kamailio to use TCP for SIP when relaying the call?  I
> haven't found much info on it.

for example, you can assign to $du a sip uri that includes ;transport=tcp
param.

-- juha

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Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

2014-12-18 Thread José Seabra

Hi Carsten


>- Do you have the Dispatcher Activity detection turned on?
>Can you send me your config, so i can verify a few things?
My dispatcher configuration is:

# - dispatcher params -
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "force_dst", 2)
modparam("dispatcher", "flags", 3)
modparam("dispatcher", "db_url", DBURL)
modparam("dispatcher", "dst_avp", "$avp(s:dst)")
modparam("dispatcher", "cnt_avp", "$avp(s:dsp_uri_count)")
modparam("dispatcher", "grp_avp", "$avp(s:dsp_uri_grp)")
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_ping_from", "sip:sipcheck@10.0.20.102:5062")
modparam("dispatcher", "ds_ping_interval", 10)
modparam("dispatcher", "ds_probing_threshhold", 3)
modparam("dispatcher", "ds_probing_mode", 2)


>Do you do something like "ds_mark_dst()" in your failure route?

Yes, I'm using ds_mark_dst inside of failure route, but the "REGISTER" is
not entering in failure route in any case, because when "REGISTER" gets the
function ds_select_dst, this function returns false and my kamailio sends a
statless reply to phone and drops the message.

I did a test that was comment "ds_mark_dst("ip");" in failure route and
even with this I still having the same issue.


route[HANDLE_REGISTER]
{

   t_set_fr(0,2000);
   t_on_failure("PRXFAILURE");

   if(!ds_select_dst("9", "4"))
   {
   xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
   sl_send_reply("503", "No proxy available, try again later");
   exit;
   }

   ...

   route(RELAY);

}


failure_route[PRXFAILURE] {
   if(is_method("REGISTER") && (t_check_status("500") || t_branch_timeout()
|| !t_branch_replied()))
   {
   $var(cache_user) = $tU + "@" + $td;
   $sht(prxcache=>$var(cache_user))=$null;
   ds_mark_dst("ip");
   if(ds_next_dst())
   {
 #again set local timeout for reply
 t_set_fr(0,2000);
 t_on_failure("PRXFAILURE");
 xlog("L_INFO", "PRX FAILURE -REGISTER New dest $du ID=$ci\n");
 $sht(tmpcache=>$var(cache_user))=$du;
 route(RELAY);
 exit;
}
else
{
  #last available node failed to reply, no other destinations
available
  send_reply("404", "No destination");
  exit;


 }

}
}

BR
José Seabra

2014-12-18 14:43 GMT+00:00 Carsten Bock :
>
> Hi,
>
> quick questions:
> - Do you have the Dispatcher Activity detection turned on?
> - Do you do something like "ds_mark_dst()" in your failure route?
> - Can you send me your config, so i can verify a few things?
>
> Thanks,
> Carsten
>
>
>
> 2014-12-18 15:14 GMT+01:00 José Seabra :
> > Hello Daniel,
> > Thank you for your reply,
> >
> > Please check out the attachment with kamailio logs in mode debug 3, i
> didn't
> > find any errors, the only thingh that is confusing me is this line:
> > /usr/local/sbin/kamailio[13690]: DEBUG: dispatcher [dispatch.c:1745]:
> > ds_select_dst_limit(): alg hash [0]
> >
> > I don't know if this "ds_select_dst_limit(): alg hash [0]" is related
> with
> > algorithm parameter in function ds_select_dst, but, if yes i cannot
> > understand why it is 0 because I'm using algorithm 4.
> >
> > BR
> > José Seabra
> >
> >
> > 2014-12-18 11:35 GMT+00:00 Daniel-Constantin Mierla :
> >>
> >> Hello,
> >>
> >> I asked in the first email - do you get error messages in syslog? If
> not,
> >> then you have to run with debug=3 and look at the log messages to see
> if you
> >> get further hints.
> >>
> >> Cheers,
> >> Daniel
> >>
> >>
> >> On 18/12/14 12:29, José Seabra wrote:
> >>
> >> Hello Daniel,
> >> Do you need more information from my kamailio settup, in order to try
> >> understand why this issue happens?
> >> Thank you
> >> BR
> >> José Seabra
> >>
> >>
> >> 2014-12-16 12:16 GMT+00:00 José Seabra :
> >>>
> >>> Update:
> >>> My last email has the wrong  example, the correct code is:
> >>>
> >>> if(!ds_select_dst("9", "4"))
> >>> {
> >>> xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
> >>> sl_send_reply("503", "No proxy available, try again later");
> >>> exit;
> >>> }
> >>>
> >>> I'm using algorithm 4.
> >>>
> >>> 2014-12-16 12:14 GMT+00:00 José Seabra :
> 
>  Hello Daniel,
> 
>  Yes it enter inside a condition.
> 
>  my code is:
> 
>  if(!ds_select_dst("9", "0"))
>  {
>  xlog("L_ERR", "Failed to select proxy - R=$ru ID=$ci\n");
>  sl_send_reply("503", "No proxy available, try again later");
>  exit;
>  }
>  and all sip registrers sent receives a sip reply 503 "No proxy
>  available, try again later"
> 
> 
>  BR
>  José Seabra
> 
>  2014-12-16 12:06 GMT+00:00 Daniel-Constantin Mierla <
> mico...@gmail.com>:
> >
> > Hello,
> >
> > what do you mean starts returning null?
> >
> > Does it enter inside a condition like:
> >
> > if(!ds_select_dst(...))
> >
> > Are there any error messages in syslog?
> >
> > Daniel
> >
> >>

[SR-Users] TLS enable false.

2014-12-18 Thread Thanh Truong

Hi all,

I have tried several configure TLS in kamailio but no luck.

Please give me some suggestion that I can make it work correctly.

This is my configure in TLS module.

modparam("tls", "tls_method", "SSLv23")

modparam("tls", "private_key", "/usr/local/etc/kamailio/ca/privkey.pem")

modparam("tls", "certificate",
"/usr/local/etc/kamailio/ca/kamailio1_cert.pem")

modparam("tls", "ca_list", "/usr/local/etc/kamailio/ca/calist.pem")

modparam("tls", "verify_certificate", 1)

modparam("tls", "require_certificate", 1)


I am only getting issue with verify_certifiate = 1, it i let it to 0, my
client can register correctly.


When I set this flag, i got message from server as:


Dec 18 10:26:31 17237 /usr/local/sbin/kamailio[12655]: ERROR: 
[tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading

Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: tls
[tls_server.c:1193]: tls_read_f(): TLS accept:error:140890C7:SSL
routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate

Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: 
[tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading


I Cant add any pem file into client, i have used Blink phone but no luck.
Thank all in advance.

ThanhTruong.
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Re: [SR-Users] TLS enable false.

2014-12-18 Thread Rob Moore

Hi Thanh,

Are you intending to use Client certificates in your setup?

If not, the  error “SSL3_GET_CLIENT_CERTIFICATE “ would lead me to believe that 
your problem is modparam("tls", "require_certificate", 1) which when enabled 
kamailio will require a certificate from the client.
I’m not an expert with TLS, but this may help depending on what type of TLS 
setup you are trying to achieve.

From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of 
Thanh Truong
Sent: 18 December 2014 15:28
To: kamailio
Subject: [SR-Users] TLS enable false.

Hi all,

I have tried several configure TLS in kamailio but no luck.

Please give me some suggestion that I can make it work correctly.

This is my configure in TLS module.

modparam("tls", "tls_method", "SSLv23")
modparam("tls", "private_key", "/usr/local/etc/kamailio/ca/privkey.pem")
modparam("tls", "certificate", "/usr/local/etc/kamailio/ca/kamailio1_cert.pem")
modparam("tls", "ca_list", "/usr/local/etc/kamailio/ca/calist.pem")
modparam("tls", "verify_certificate", 1)
modparam("tls", "require_certificate", 1)

I am only getting issue with verify_certifiate = 1, it i let it to 0, my client 
can register correctly.

When I set this flag, i got message from server as:

Dec 18 10:26:31 17237 /usr/local/sbin/kamailio[12655]: ERROR:  
[tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading
Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: tls 
[tls_server.c:1193]: tls_read_f(): TLS accept:error:140890C7:SSL 
routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate
Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR:  
[tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading

I Cant add any pem file into client, i have used Blink phone but no luck.
Thank all in advance.

ThanhTruong.

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Re: [SR-Users] TLS enable false.

2014-12-18 Thread Thanh Truong

Hi Rob Moore,

Yes, I have intended to use TLS in client side to verify with server side.

I have tried to create cert files as :
Quick Certificate Howto
in http://kamailio.org/docs/modules/stable/modules/tls.html#tls.debugging

Then, I tried to use Blink phone to user crt file, But I see that I cant
add any pem file to this.

Can you suggest ?

Thank all,
ThanhTruong

Thanks,
ThanhTruong.


On Thu, Dec 18, 2014 at 11:28 PM, Rob Moore  wrote:
>
>  Hi Thanh,
>
>
>
> Are you intending to use Client certificates in your setup?
>
>
>
> If not, the  error “SSL3_GET_CLIENT_CERTIFICATE “ would lead me to
> believe that your problem is modparam("tls", "require_certificate", 1) which
> when enabled kamailio will require a certificate from the client.
>
> I’m not an expert with TLS, but this may help depending on what type of
> TLS setup you are trying to achieve.
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.sip-router.org] *On
> Behalf Of *Thanh Truong
> *Sent:* 18 December 2014 15:28
> *To:* kamailio
> *Subject:* [SR-Users] TLS enable false.
>
>
>
> Hi all,
>
>
>
> I have tried several configure TLS in kamailio but no luck.
>
>
>
> Please give me some suggestion that I can make it work correctly.
>
>
>
> This is my configure in TLS module.
>
>
>
> modparam("tls", "tls_method", "SSLv23")
>
> modparam("tls", "private_key", "/usr/local/etc/kamailio/ca/privkey.pem")
>
> modparam("tls", "certificate",
> "/usr/local/etc/kamailio/ca/kamailio1_cert.pem")
>
> modparam("tls", "ca_list", "/usr/local/etc/kamailio/ca/calist.pem")
>
> modparam("tls", "verify_certificate", 1)
>
> modparam("tls", "require_certificate", 1)
>
>
>
> I am only getting issue with verify_certifiate = 1, it i let it to 0, my
> client can register correctly.
>
>
>
> When I set this flag, i got message from server as:
>
>
>
> Dec 18 10:26:31 17237 /usr/local/sbin/kamailio[12655]: ERROR: 
> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading
>
> Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: tls
> [tls_server.c:1193]: tls_read_f(): TLS accept:error:140890C7:SSL
> routines:SSL3_GET_CLIENT_CERTIFICATE:peer did not return a certificate
>
> Dec 18 10:26:46 17237 /usr/local/sbin/kamailio[12656]: ERROR: 
> [tcp_read.c:1279]: tcp_read_req(): ERROR: tcp_read_req: error reading
>
>
>
> I Cant add any pem file into client, i have used Blink phone but no luck.
>
> Thank all in advance.
>
>
>
> ThanhTruong.
>
>
>
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>
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[SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin

Hi!
I need to establish calls between WebRTC and usual SIP clients
(exactly, sipml/jssip and linphone-android).
I used configs from https://github.com/caruizdiaz/kamailio-ws and
latest git master HEAD of both kamailio and
rtpengine. I got calls from webrtc to android working correctly (but only with
Firefox browser), even with video. But in other directions i have some
issues because of lack of RTP delivery or RTP timeouts.
I have some logs to show you regarding this:
https://gist.github.com/krieger-od/27c6f3e4924f5e21352e (works),
https://gist.github.com/krieger-od/196bcfbd331d621427ef (doesn't
work).
I would really love to get some quick help from anyone. For direct
manual fixing, I can give a couple of hundreds of bucks.

Looking forward impatiently for reply from anyone having something to say.

-- 
Andrey Utkin

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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Richard Fuchs

On 12/18/14 12:11, Andrey Utkin wrote:
> Hi!
> I need to establish calls between WebRTC and usual SIP clients
> (exactly, sipml/jssip and linphone-android).
> I used configs from https://github.com/caruizdiaz/kamailio-ws and
> latest git master HEAD of both kamailio and
> rtpengine. I got calls from webrtc to android working correctly (but only with
> Firefox browser), even with video. But in other directions i have some
> issues because of lack of RTP delivery or RTP timeouts.
> I have some logs to show you regarding this:
> https://gist.github.com/krieger-od/27c6f3e4924f5e21352e (works),
> https://gist.github.com/krieger-od/196bcfbd331d621427ef (doesn't
> work).
> I would really love to get some quick help from anyone. For direct
> manual fixing, I can give a couple of hundreds of bucks.
> 
> Looking forward impatiently for reply from anyone having something to say.

Write error on RTP socket usually indicates an incorrect network setup,
for example trying to use a source address for IP packets which isn't
bound to any local network interface (especially if you're sitting
behind NAT), or local iptables rules rejecting outgoing IP packets, or
missing IP routes to the destination. Perhaps post your network setup
and the CLI arguments to rtpengine you're using.

cheers

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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin

2014-12-18 19:30 GMT+02:00 Richard Fuchs :
> Write error on RTP socket usually indicates an incorrect network setup,
> for example trying to use a source address for IP packets which isn't
> bound to any local network interface (especially if you're sitting
> behind NAT), or local iptables rules rejecting outgoing IP packets, or
> missing IP routes to the destination. Perhaps post your network setup
> and the CLI arguments to rtpengine you're using.

I use VPS from DigitalOcean. Likely no NAT. Only one public IP, and
one domain name used (whdd.org). Exactly the same situation was with
Amazon VPS behind amazonish NAT.
My configs are here: https://github.com/krieger-od/webrtc_kamailio_etc
(only kamailio.cfg and modules.cfg are used, kamailio*.cfg are not).
Also there you can check command line used to run rtpengine. To put it
there, it is
/home/krieger/rtpengine/daemon/rtpengine --interface=188.226.241.236
--listen-ng=127.0.0.1:2 -m 3 -M 35000 --foreground
--log-stderr


-- 
Andrey Utkin

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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Richard Fuchs

On 12/18/14 12:55, Andrey Utkin wrote:
> 2014-12-18 19:30 GMT+02:00 Richard Fuchs :
>> Write error on RTP socket usually indicates an incorrect network setup,
>> for example trying to use a source address for IP packets which isn't
>> bound to any local network interface (especially if you're sitting
>> behind NAT), or local iptables rules rejecting outgoing IP packets, or
>> missing IP routes to the destination. Perhaps post your network setup
>> and the CLI arguments to rtpengine you're using.
> 
> I use VPS from DigitalOcean. Likely no NAT. Only one public IP, and
> one domain name used (whdd.org). Exactly the same situation was with
> Amazon VPS behind amazonish NAT.
> My configs are here: https://github.com/krieger-od/webrtc_kamailio_etc
> (only kamailio.cfg and modules.cfg are used, kamailio*.cfg are not).
> Also there you can check command line used to run rtpengine. To put it
> there, it is
> /home/krieger/rtpengine/daemon/rtpengine --interface=188.226.241.236
> --listen-ng=127.0.0.1:2 -m 3 -M 35000 --foreground
> --log-stderr

Amazon NAT is exactly why I've mentioned it, because on an Amazon
system, if you don't use the --interface option correctly
($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
that show in your log.

But these configs and this CLI line don't match the logs you posted
earlier. Is it also write errors you're getting with those?

cheers

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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin

2014-12-18 20:05 GMT+02:00 Richard Fuchs :
> Amazon NAT is exactly why I've mentioned it, because on an Amazon
> system, if you don't use the --interface option correctly
> ($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
> that show in your log.

Thank you, will try with such notation on amazon host now.

> But these configs and this CLI line don't match the logs you posted
> earlier. Is it also write errors you're getting with those?

Will provide you shortly with logs from DigitalOcean VPS instance,
configs for which i've published on github.

-- 
Andrey Utkin

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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin

2014-12-18 20:09 GMT+02:00 Andrey Utkin :
> 2014-12-18 20:05 GMT+02:00 Richard Fuchs :
>> Amazon NAT is exactly why I've mentioned it, because on an Amazon
>> system, if you don't use the --interface option correctly
>> ($INT_IP!$EXT_IP notation), you get exactly these kinds of write errors
>> that show in your log.
>
> Thank you, will try with such notation on amazon host now.

This alone haven't helped with Amazon instance.
Will make tests from DigitalOcean instance and provide you with logs.

-- 
Andrey Utkin

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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin

This works: call from sipml to linphone android:
rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3


This doesn't work: few seconds after answer, there's no media from
remote subscriber, then linphone android agent hangs up; sipml hangs
pretending being in call.
rtpengine: https://gist.github.com/krieger-od/9eb120199ec99d1adcb4
kamailio: https://gist.github.com/krieger-od/26060c8d1d657458d9d2
ngrep: https://gist.github.com/krieger-od/d677864fcab8c508adde

-- 
Andrey Utkin

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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin

2014-12-18 20:38 GMT+02:00 Andrey Utkin :
> This works: call from sipml to linphone android:
> rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
> kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
> ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
>
>
> This doesn't work: few seconds after answer, there's no media from
> remote subscriber, then linphone android agent hangs up; sipml hangs
> pretending being in call.
> rtpengine: https://gist.github.com/krieger-od/9eb120199ec99d1adcb4
> kamailio: https://gist.github.com/krieger-od/26060c8d1d657458d9d2
> ngrep: https://gist.github.com/krieger-od/d677864fcab8c508adde

Doesn't work in different way:
jssip web agent calls to android linphone: call is connected, but
android doesn't see peer's video, and jssip doesn't have peer's audio:
rtpengine: https://gist.github.com/krieger-od/cd63ef99d06212b30379
kamailio: https://gist.github.com/krieger-od/55c4b5785c5821315955
ngrep: https://gist.github.com/krieger-od/650575c0f96285d78d17

-- 
Andrey Utkin

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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Andrey Utkin

Also I encounter such issue: even in the working scenario, the hangup
of one peer doesn't make the call end for another peer.
rtpengine: https://gist.github.com/krieger-od/1cfe84b53dc0d29cfb90
kamailio: https://gist.github.com/krieger-od/11c6bbf7dad15382e81b
ngrep: https://gist.github.com/krieger-od/ec6c21c6f9dde1400578

Also, this deploy of http://vmwrtc.intersog.com/sipml5/call.htm works
ok, and official one http://sipml5.org/call.htm?svn=222 has more
issues.


-- 
Andrey Utkin

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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Richard Fuchs

On 12/18/14 13:38, Andrey Utkin wrote:
> This works: call from sipml to linphone android:
> rtpengine: https://gist.github.com/krieger-od/bf8503fe7643c0571b58
> kamailio: https://gist.github.com/krieger-od/c119d64af6edcde3fc46
> ngrep: https://gist.github.com/krieger-od/cb5829be7a55a7acf9d3
> 
> 
> This doesn't work: few seconds after answer, there's no media from
> remote subscriber, then linphone android agent hangs up; sipml hangs
> pretending being in call.
> rtpengine: https://gist.github.com/krieger-od/9eb120199ec99d1adcb4
> kamailio: https://gist.github.com/krieger-od/26060c8d1d657458d9d2
> ngrep: https://gist.github.com/krieger-od/d677864fcab8c508adde

Couple of things to try here...

In the second case, Firefox kept sending answers (200 OK) for no good
reason. Check Firefox's debug log / JS console / whatever if there's any
error or warnings messages.

Try running rtpengine with the --dtls-passive switch, even though that
shouldn't make a difference after ICE went through (which it does).

Add rtcp-mux=demux to your rtpengine flags in your Kamailio config,
should be safe to use everywhere.

It's curious that Firefox was using RTP/SAVPF as protocol in one call
and UDP/TLS/RTP/SAVPF in another. I don't suspect that this would cause
a problem, but it's worth trying to specify the protocol not as
RTP/SAVPF but rather as transport-protocol=UDP/TLS/RTP/SAVPF. Note that
in the case of answers, you don't need to specify any protocol at all.

In the third case, the audio stream was setup as a=sendonly, explaining
the one-way audio. Probably caused by Firefox not being able to access
the playback device.

Last but not least, try a different browser, either a newer version of
Firefox (it's up to v34 I see), or Chrome.

As for the call teardown, it's Kamailio's responsibility (through the
config) to let rtpengine know that the call has been terminated, for
example when handling the BYE method. Otherwise you'll see timeouts in
rtpengine. Even though the BYE should still be relayed to the other SIP
end, causing the call to be hung up there.

cheers

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Re: [SR-Users] Can't start Kamailio with both db_postgres and tls

2014-12-18 Thread James Cloos

> "DM" == Daniel-Constantin Mierla  writes:

DM> The question would be more specific to the error message printed from
DM> postgres client library:

DM> FATAL:  no pg_hba.conf entry for host "129.240.1.1", user
DM> "foo_test_user", database " foo_test", SSL off

DM> Is it something that is documented somewhere or maybe some web search
DM> can indicate the reasons why it happens?

His initial post mentioned that they require ssl for the pg tcp sockets;
the error about pg_hba just confirms that.

-JimC
-- 
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Re: [SR-Users] SIP Fragments

2014-12-18 Thread James Cloos

> "MS" == Marc Soda  writes:

MS> I'm having a problem reassembling UDP packets on my Asterisk servers after
MS> passing through Kamailio

You could try having the kama->ast socket use tcp.

-JimC
-- 
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Re: [SR-Users] sr-users Digest, Vol 115, Issue 16

2014-12-18 Thread Kees Meijs - SIGNET B.V.

Hi there,

We haven't used gdb on Kamailio yet but some time soon we'll most likely
will. Obviously I'll post our findings.

Thanks again!

Cheers,
Kees

On 17-12-14 12:00, sr-users-requ...@lists.sip-router.org wrote:
> did you get a core dump file? If yes, send the full backtrace taken with
> gdb.


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Re: [SR-Users] webrtc clients support using rtpengine

2014-12-18 Thread Juha Heinanen

Richard Fuchs writes:

> In the third case, the audio stream was setup as a=sendonly, explaining
> the one-way audio. Probably caused by Firefox not being able to access
> the playback device.

as i reported in another message, i have also lately noticed in my tests
that firefox sends invite with a=sendonly attribute.  still don't know
why, because the same firefox can play audio without problems using the
html  tag.

-- juha

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Re: [SR-Users] Kamailio CDRs

Re: [SR-Users] SIP Fragments

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Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

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Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

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Re: [SR-Users] kamailio dispatcher module stops processing REGISTER

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Re: [SR-Users] TLS enable false.

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[SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] webrtc clients support using rtpengine

Re: [SR-Users] Can't start Kamailio with both db_postgres and tls

Re: [SR-Users] SIP Fragments

Re: [SR-Users] sr-users Digest, Vol 115, Issue 16

Re: [SR-Users] webrtc clients support using rtpengine

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