[OpenSIPS-Users] Use Gstreamer RTP packets as source

Bogdan-Andrei Iancu bogdan at opensips.org
Thu Aug 3 06:02:42 EDT 2017


Hi Michael,

There is some mechanism (old, so not something fancy) that may help you 
in doing this.

When receiving the INVITE, create the transaction (t_newtran()), 
eventually sent a 100 trying reply back and then use t_write_unix() to 
dump the info on INVITE to an external app:
http://www.opensips.org/html/docs/modules/2.3.x/tm.html#idp5798928
after that, from script, do not do any SIP signaling for the INVITE.

Your external app is responsible to construct the SDP and the 200 OK and 
to use the MI t_reply function to make OpenSIPS to send it back to caller:
http://www.opensips.org/html/docs/modules/2.3.x/tm.html#idp5848144

Will this work for you ?

Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com

OpenSIPS Bootcamp 2017, Houston, US
   http://opensips.org/training/OpenSIPS_Bootcamp_2017.html

On 08/03/2017 12:21 AM, Michael Smith wrote:
>
>
>
> Hello,
>
> Yes! I want to provide the RTP packets via gstreamer and send them 
> with OpenSIPS to another host that will decode the RTP packets. Do you 
> know how can I do it?
>
> Thanks!
> ------------------------------------------------------------------------
> *De:* Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Enviado:* quarta-feira, 2 de agosto de 2017 11:57:36
> *Para:* OpenSIPS users mailling list; Michael Smith
> *Assunto:* Re: [OpenSIPS-Users] Use Gstreamer RTP packets as source
> Hi Michael,
>
> Do you want to have OpenSIPS acting as an UAS end point for SIP while 
> providing the RTP via gstreamer ?
>
> Regards,
> Bogdan-Andrei Iancu
>    OpenSIPS Founder and Developer
>    http://www.opensips-solutions.com
>
> OpenSIPS Bootcamp 2017, Houston, US
>    http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
> On 07/30/2017 09:01 PM, Michael Smith wrote:
>>
>>
>> Hello,
>>
>>
>> I need to stream audio in many different encoding algorithms (G711, 
>> G722, MPEG4, etc) and I thought to use Gstreamer to encode the audio 
>> and OpenSIPS to send using a SIP communication. Will this work? Can I 
>> send the RTP encoded packets over a SIP communication using OpenSIPS?
>>
>>
>> Sorry for the rookie question.
>>
>>
>> Any tip will be very helpful,
>>
>> Thanks
>>
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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