I am looking for expert advise on dynamic resource allocation in VoIP...
I am trying to come up with some idea for using of dynamic badwidth adaptations for SIP teleconferencing in mobile wireless net? Would it be possible to use RTCP messagas to communicate resource availability ( bandwidth / jitter) so that dynamic adaptations can be initiated ( for example: stop sending video but keep audio, if bandwidth cannot support both...)
Is there any dynamic resource allocation protocol extension available for SIP?
Is it possible to handle rtp/ rtcp in H323 ?