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.sa...@enukesoftware.com
<mailto:abhishek.sa...@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
<mico...@gmail.com <mailto:mico...@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:ad...@abc.com
<mailto:sip%3aad...@abc.com>>;tag=bzhwwG8nT2gFwwJgIyrz.
To: <sip:h...@abc.com <mailto:sip%3ah...@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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