Hi,

I have reported a bug on rptengine github, for the crash issue: https://github.com/sipwise/rtpengine/issues/27

You mentioned that you have been using rtpengine kamailio module and the rtpengine debian package with success. Was it on ubuntu box or some other linux system? Sorry for asking this, but i am not able move ahead because of this. Any other module suggestions from your end?

Thanks 

 

On Wed, Sep 17, 2014 at 6:45 PM, Abhishek Saini <abhishek.saini@enukesoftware.com> wrote:
Hi Daniel,

Here is something i traced in the log:

ip-172-31-47-138 rtpengine[4879]: Unknown flag encountered: 'force'
ip-172-31-47-138 kernel: [4155571.651074] traps: rtpengine[4884] general protection ip:41e313 sp:7f2bf1934418 error:0 in rtpengine[400000+30000]

What's the cause of this error? i am using code from the master branch. Perhaps this has something to do with the rptengine service crash/termination.

Regards

On Wed, Sep 17, 2014 at 1:28 PM, Abhishek Saini <abhishek.saini@enukesoftware.com> wrote:
Hi 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.

I also updated this param : modparam("nathelper", "sipping_from", "sip:pinger@abc.com") to my domain

Now the scenario is as follows:

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.

2) Still when i call from webrtc to iphone - the retpengine service of ubuntu terminates/crashes (like before) and needs to be restarted.

Does it have any thing to do with rtp port ranges? or is there some other misconfiguration?


Regards,
Abhishek

 


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:
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



-- 
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