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

Wed Sep 17 09:58:09 CEST 2014

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 at 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 at 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 at 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 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
>>>
>>>
>>
>
> --
> 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
>
>
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