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
<mico...@gmail.com <mailto:mico...@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
<mico...@gmail.com <mailto:mico...@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@lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
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--
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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sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
