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<tt>Hi Michael,<br>
<br>
There is some mechanism (old, so not something fancy) that may
help you in doing this.<br>
<br>
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:<br>
<a class="moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/2.3.x/tm.html#idp5798928">http://www.opensips.org/html/docs/modules/2.3.x/tm.html#idp5798928</a><br>
after that, from script, do not do any SIP signaling for the
INVITE.<br>
<br>
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:<br>
<a class="moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/2.3.x/tm.html#idp5848144">http://www.opensips.org/html/docs/modules/2.3.x/tm.html#idp5848144</a><br>
<br>
Will this work for you ?<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html">http://opensips.org/training/OpenSIPS_Bootcamp_2017.html</a>
</pre>
<div class="moz-cite-prefix">On 08/03/2017 12:21 AM, Michael Smith
wrote:<br>
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Hello,</div>
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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?</div>
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Thanks!</div>
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<hr style="display:inline-block;width:98%" tabindex="-1">
<div id="divRplyFwdMsg" dir="ltr"><font style="font-size:11pt"
color="#000000" face="Calibri, sans-serif"><b>De:</b>
Bogdan-Andrei Iancu <a class="moz-txt-link-rfc2396E" href="mailto:bogdan@opensips.org"><bogdan@opensips.org></a><br>
<b>Enviado:</b> quarta-feira, 2 de agosto de 2017 11:57:36<br>
<b>Para:</b> OpenSIPS users mailling list; Michael Smith<br>
<b>Assunto:</b> Re: [OpenSIPS-Users] Use Gstreamer RTP packets
as source</font>
<div> </div>
</div>
<div><tt>Hi Michael,<br>
<br>
Do you want to have OpenSIPS acting as an UAS end point for
SIP while providing the RTP via gstreamer ?<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com" moz-do-not-send="true">http://www.opensips-solutions.com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html" moz-do-not-send="true">http://opensips.org/training/OpenSIPS_Bootcamp_2017.html</a>
</pre>
<div class="moz-cite-prefix">On 07/30/2017 09:01 PM, Michael
Smith wrote:<br>
</div>
<blockquote type="cite"
cite="mid:RO1P215MB0393EFED39BEDD287C0D2485CFBD0@RO1P215MB0393.LAMP215.PROD.OUTLOOK.COM">
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<p>Hello,</p>
<p><br>
</p>
<p>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?</p>
<p><br>
</p>
<p>Sorry for the rookie question.</p>
<p><br>
</p>
<p>Any tip will be very helpful,</p>
<p>Thanks</p>
<p><br>
</p>
</div>
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