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 G722, G711u/a and so on.

... " container formats: asf, avi, 3gp/mp4/mov, flv, mpeg-ps/ts, mkv/webm, mxf, ogg"

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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