Hello,
i don't think that GStreamer can handle SIP or SDP information.
It is just a ...
" GStreamer can bridge to other multimedia frameworks in order to reuse
existing components (e.g. codecs) and use platform input/output mechanisms:"
And to point at RTP ... some codes are missing like
Hello,
I am using Kamailio as registration server and FreeSwitch for signalling
(RTP packet handling). Can I use GStreamer instead of Freeswitch or
Asterisk?
Thank you.
Regards,
CM
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