[SR-Users] No audio/video transmission over different networks
miconda at gmail.com
Thu Sep 4 14:32:06 CEST 2014
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
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
> On Thu, Sep 4, 2014 at 1:28 PM, Daniel-Constantin Mierla
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
> 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
> 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.
>> On Thu, Sep 4, 2014 at 12:34 PM, Daniel-Constantin Mierla
>> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>> 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
>> On 03/09/14 12:23, Abhishek Saini wrote:
>> 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/#%21/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>
> 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
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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