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

Abhishek Saini abhishek.saini at enukesoftware.com
Tue Sep 16 09:45:10 CEST 2014


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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20140916/ad429b2b/attachment.html>


More information about the sr-users mailing list