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@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@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://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany


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sr-users@lists.sip-router.org
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-- 
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

-- 
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