4 Sep
2014
4 Sep
'14
4:24 p.m.
Hi,
Please find attached the output of ngrep for three type of
combinations/connections:
key: Blink is the desktop sip client and ntw means network.
blink2blink_same_ntw_successful
webrtc2blink_same_ntw_failed
webrtc2webrtc_same_ntw_successful
We also need to enable webrtc to classic sip phone calls, like on
iphones/desktops etc. I could not find a good tutorial on rtpengine, and
the steps to replace rtpproxy with rtpengine.
Please suggest me on this.
Regards,
Abhishek
On Thu, Sep 4, 2014 at 6:02 PM, Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
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(a)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(a)gmail.com> wrote:
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
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:
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Next Kamailio Advanced Trainings 2014 - http://www.asipto.com
Sep 22-25, Berlin, Germany
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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