[SR-Users] SDP IPv4 has been concatenate - no audio

Loic Chabert chabert.loic.74 at gmail.com
Fri Aug 7 16:40:01 CEST 2015


Hello,

I have set on the right place "route(RTPPROXY")", and now it works for
internal calls and external calls.
Reason: my request passing througt RTPPROXY twice ...

One last problem:
- 102 initiate a call to 101
- 101 refuse call with a 486 response
- as asterisk dialplan said: launch voicemail app
- Sounds files has been read from asterisk, but after 5 secondes, session
has been cut with a BYE request sent by Asterisk.

Please find in attachement pcap trace file (91.x.x.x is wan kamailio
interface, 10.0.247.197 is lan kamailio interface, facing to asterisk
cluster)

Why asterisk send this BYE ? Kamailio does not force him to send this BYE...

Thanks,
Loic.


2015-08-07 10:34 GMT+02:00 Daniel-Constantin Mierla <miconda at gmail.com>:

> Hello,
>
> look at the sip traffic and see what is in SDP, if you don't get audio,
> maybe the other ip is advertised.
>
> Cheers,
> Daniel
>
>
> On 07/08/15 09:16, Loic Chabert wrote:
>
> Hello Daniel,
>
> I have changed my rtpproxy by rtpengine. I have explicitly define public
> and private interfaces, and now it work as expected for external calls
> (througth PSTN).
> But for now, after this change, internal call (like 100 call 101), does
> not work any more.
>
> I need more investigation to see what append on my call flow.
>
> I will update you asap.
>
> Thanks,
> Regards.
>
>
> 2015-08-07 9:04 GMT+02:00 Daniel-Constantin Mierla <miconda at gmail.com>:
>
>> Hello,
>>
>> On 30/07/15 17:38, Loic Chabert wrote:
>>
>> Hello everyone,
>>
>> I'm trying put kamailio in front of asterisk server farm. Fow now, 2
>> asterisk servers are running and i'm trying to make some basic calls
>> between two UACc.
>>
>> All asterisk servers has been ofuscaded from public internet using
>> 10.189.122.0/24 network.
>> All trafic must be passed throught asterisk so RTPproxy is used to (and
>> used for rtp bridging).
>> Kamailio and rtpproxy is running with public IP address, and private ip
>> address (mhomed=1)
>>
>> But a wired thing append on my SDP body: c line have two rtpproxy public
>> addresses concatenate (see my capture attached).
>>
>> Any reason for this ? Only invite method from my asterisk contains 2
>> publics IP addresses concatenated.
>>
>> Does it mean than rtp_manage as been executed twice ?
>>
>> It could be that it was executed twice. As pointed in another response,
>> look at what is received on the network and in the logs.
>>
>> You can enable cfgtrace for debugger module in order to see what actions
>> are executed from configuration files -- it is good to spot quickly errors
>> in the logic of config file.
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Book: SIP Routing With Kamailio - http://www.asipto.com
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> --
> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
> Book: SIP Routing With Kamailio - http://www.asipto.com
>
>
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