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

Wed Sep 17 15:15:13 CEST 2014

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 at 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 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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