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

[Daniel-Constantin Mierla]

Hello,

maybe you should play with kamailio master branch (which is in testing 
phase before becoming 4.2)  -- there you have the rtpengine -- and see 
if you get it working. Once that, you can look at using an older 
version, knowing you have it working and be able to compare. As I needed 
latest features, whenever I needed webrtc gatewaying, I used devel 
branch of rtpengine module.

Cheers,
Daniel

On 16/09/14 14:24, Abhishek Saini wrote:
> Hi Daniel,
>
>
> I was able to solve a fraction of my problem, Actually, the github 
> link had used rtpengine.so and i was using rptproxy-ng.so, there is a 
> difference in the flag conventions between the two; i modified that to 
> achieve a little progress.
>
> Now, i am able to call on webrtc(firefox) from sip phone. However, 
> after accepting call, there is no audio, and disconnecting the call 
> from either end does not disconnect the call.
>
> When i try to call from webrtc(firefox) to sip phone, there is no 
> signalling at all, and the sip phone to webrtc calls can't connect 
> after that. (I analyzed that mediaproxy-ng/rtpengine process 
> terminates and has to be started again)
>
> Following are the links to my latest kamailio.cfg file and port trace 
> log of sip messages.
> http://jmp.sh/o0apKgP
> http://jmp.sh/HXnFRQj
>
> I am clueless at the moment!
>
> Regards,
> Abhishek
>
>
>
> On Tue, Sep 16, 2014 at 1:15 PM, Abhishek Saini 
> <abhishek.saini at enukesoftware.com 
> <mailto:abhishek.saini at enukesoftware.com>> wrote:
>
>     Hi Daniel,
>
>     Thanks for this.
>
>     I took the entire config files and configured it as per my ips and
>     ports, after doing that, still no call establishment(webrtc to
>     classic sip phones and vice-versa). Following is what i get in
>     kamailio.log:
>
>     rtpp_test(): rtp proxy <udp:127.0.0.1:7722
>     <http://127.0.0.1:7722>> found, support for it enabled
>     ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call():
>     unknown option ` '
>     ERROR: <script>: ==>
>     duri=[sip:nudg.com:5060;lr;sipml5-outbound;transport=tcp]
>     INFO: <script>: Request coming from WS
>     ERROR: rtpproxy-ng [rtpproxy.c:1254]: rtpp_function_call():
>     unknown option ` '
>     INFO: <script>: Reply from softphone: 100
>
>     And this SIP message:
>     SIP/2.0 603 Failed to get local SDP.
>
>     Regards,
>     Abhishek
>
>
>
>
>     On Mon, Sep 15, 2014 at 6:19 PM, Daniel-Constantin Mierla
>     <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>         Hello,
>
>         the reply code indicates that the media type is not supported,
>         thus there has been no gatewaying between webrtc and classic
>         rtp. Just replacing rtpproxy with rtpengine is not enough,
>         there are different parameters that have to be provided.
>
>         Searching on web, I see that Carlos has published a config for
>         it, see:
>         - https://github.com/caruizdiaz/kamailio-ws
>
>         Cheers,
>         Daniel
>
>
>         On 15/09/14 12:58, Abhishek Saini wrote:
>>         Hi,
>>
>>         I have successfully setup rtpproxy-ng kamailio module and
>>         mediaproxy-ng package on my ubuntu box. As suggested here:
>>         http://kamailio.org/docs/modules/devel/modules/rtpproxy-ng.html
>>
>>         I have kept rtpproxy-ng's configuration same as the rtpproxy
>>         module, but still not able to connect the webrtc calls to
>>         classic sip phones (and vice-versa). Below is the sip message
>>         that is traced:
>>
>>
>>         SIP/2.0 488 Not acceptable here.
>>         Via: SIP/2.0/TCP
>>         54.191.193.xxx:5060;branch=z9hG4bK6745.f449086ab0b221d6173373c$
>>         Via: SIP/2.0/WS
>>         df7jal23ls0d.invalid;received=203.92.41.2;branch=z9hG4bKExDPMNb$
>>         From: "admin" <sip:admin at abc.com
>>         <mailto:sip%3Aadmin at abc.com>>;tag=bzhwwG8nT2gFwwJgIyrz.
>>         To: <sip:hari at abc.com <mailto:sip%3Ahari at abc.com>>;tag=OIllTQf.
>>         Call-ID: 31464f04-27e6-b11c-3a63-ba1d4d2d4d5a.
>>         CSeq: 65463 INVITE.
>>         User-Agent: LinphoneIPhone/2.2.1 (belle-sip/1.3.2).
>>         Supported: replaces, outbound.
>>         Content-Length: 0.
>>
>>         Can you please let me know, what's going wrong and how can i
>>         proceed.
>>
>>         Regards,
>>         Abhishek
>>
>>
>>
>>
>
>         -- 
>         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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