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

[Abhishek Saini]

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> 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> 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> 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>;tag=bzhwwG8nT2gFwwJgIyrz.
>> To: <sip:hari 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 Mierlahttp://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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