[SR-Users] No audio/video transmission over different networks

[Daniel-Constantin Mierla]

Hello,

maybe you can send to mailing list the output of ngrep so we can look 
and check if a rtp relay is used.

If you need to bridge webrtc to classic sip phone, you have to use 
rtpengine.

Cheers,
Daniel

On 04/09/14 13:01, Abhishek Saini wrote:
> Hi Daniel,
>
> Thanks, i was able to use the command you provided, but did not find 
> the chunks you have specified(a=nortproxy:yes (iirc)) in the data. 
> Checked by calling from webrtc client to a desktop client(blink).
>
> When is rtpproxy used though? Kamailio says that it only transmits SIP 
> signals and has not much to do with the media(voice or video). So, 
> that means, it utilizes the rtpproxy to transmit the SIP signals(for 
> non-symmetric NAT), If so then i think, the rtpproxy is working fine, 
> as i have always been able to make and receive calls and only the 
> media (voice or video) are not working (cross network).
>
> I have also setup webrtc - it's working fine (firefox to firefox) but 
> when i call from firefox to desktop client, it does not work(only 
> rings, but does not connect).
> I read about webrtc_breaker but there does not seem to be a module for 
> that in kamailio.
>
> I think these two issues are somehow interlinked, please suggest me on 
> this.
>
> Regards,
> Abhishek
>
>
> On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla 
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>
>     On 04/09/14 09:20, Abhishek Saini wrote:
>>     Hi Daniel,
>>
>>     Thanks for reply.
>>
>>     I did install patched rtpproxy and did configure it the way you
>>     have described (advertising address - found that after posting
>>     the comment). But it still does not seem to work.
>>
>>     I don't quite know how can i debug, if rtpproxy is actually being
>>     used.
>     use ngrep to look at sip traffic, like:
>
>     ngrep -d any -qt -W byline port 5060
>
>     If rtpproxy was enforced, you should see a=nortproxy:yes (iirc) in
>     the SDP. Also, the media IP in SDP should change from incoming
>     INVITE to what is sent out in the IP of rtpproxy.
>
>     Cheers,
>     Daniel
>
>
>>
>>     Regards,
>>     Abhishek
>>
>>
>>
>>     On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla
>>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>>
>>         Hello,
>>
>>         no time to look at config, but if you run the sip server on a
>>         private IP behind a port forwarding address, you have to use
>>         also rtpproxy with advertising address -- see the second
>>         parameter of rtpproxy_manage() or search on the web for a
>>         patch to rtpproxy to add advertising address via command line
>>         parameter.
>>
>>         Cheers,
>>         Daniel
>>
>>
>>         On 03/09/14 12:23, Abhishek Saini wrote:
>>
>>             Hi,
>>
>>             I have setup kamailio 4.1.0 on an EC2 xlarge instance.
>>             The voice and video calls seem to work well when both the
>>             devices are connected to the same network, however, when
>>             one device connects to a different network (the two
>>             devices now are on different networks), they are able to
>>             register on SIP server, and even call can be triggered
>>             and accepted between the two devices but there is no
>>             video/audio transmission.
>>
>>             I have setup rtpproxy but i don't know whether it's
>>             working or not.
>>
>>             Any help on this would be highly appreciated.
>>
>>
>>             Following is my kamailio.cfg file:
>>
>>
>>         -- 
>>         Daniel-Constantin Mierla
>>         http://twitter.com/#!/miconda
>>         <http://twitter.com/#%21/miconda> -
>>         http://www.linkedin.com/in/miconda
>>         Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
>>         Sep 22-25, Berlin, Germany
>>
>>
>>         _______________________________________________
>>         SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
>>         mailing list
>>         sr-users at lists.sip-router.org
>>         <mailto:sr-users at lists.sip-router.org>
>>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>     -- 
>     Daniel-Constantin Mierla
>     http://twitter.com/#!/miconda  <http://twitter.com/#%21/miconda>  -http://www.linkedin.com/in/miconda
>     Next Kamailio Advanced Trainings 2014 -http://www.asipto.com
>     Sep 22-25, Berlin, Germany
>
>

-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany

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