RFC 3551 has defaults for common codecs in RTP. It is a good idea to stick to
those defaults when you can for improved interoperability. It does however
note:
4.3 Guidelines for Sample-Based Audio Encodings
An RTP audio packet
may contain any number of audio samples, subject to the constr
After exhaustive searching, I have been unable to find any standards for
interworking ISDN supplementary services to SIP (Call Forwarding, Transfer,
3-Way calling, Call Waiting, Hold, etc. as described in Q.932 and the Q.730
series and Q.950 series docs as well as other docs covering NI-2, 5ESS
