AbhishekDoes it have any thing to do with rtp port ranges? or is there some other misconfiguration?2) Still when i call from webrtc to iphone - the retpengine service of ubuntu terminates/crashes (like before) and needs to be restarted.1) I am able to call webrtc(firefox and chrome) from iphone, the signalling seems to be working fine, call can be paused, resumed etc.., but there is no audio/video transmission.Now the scenario is as follows:I also updated this param : modparam("nathelper", "sipping_from", "sip:pinger@abc.com") to my domainHi Daniel,As you instructed, i installed kamailio from the master branch (which has rtpengine module). Along with this, i installed the rtpengine package from sipwise, as instructed by them.
Regards,
On Tue, Sep 16, 2014 at 6:31 PM, Daniel-Constantin Mierla <miconda@gmail.com> wrote: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:
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)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.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.
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:
AbhishekAnd this SIP message: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:Hi Daniel,Thanks for this.
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
SIP/2.0 603 Failed to get local SDP.
Regards,
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
-- 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