[OpenSIPS-Users] Use Gstreamer RTP packets as source

[Michael Smith]

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

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