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@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@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@abc.com>;tag=bzhwwG8nT2gFwwJgIyrz.
To: <sip:hari@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://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany