SIP resource allocation?

5 pts.
Tags:
RTCP
SIP
VoIP resource allocation
Hello, 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 ? Thanks -vl

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Thanks. We'll let you know when a new response is added.

Yes, through RTP/RTCP the bandwidth can be controlled/allocated dynamically. But there need a interface for communicating with the RTP/RTCP.

H.323 and SIP also provides an interface for supporting RTP/RTCP.
H.323 uses H.245 signaling protocal for handling RTP/RTCP. SIP uses SDP as interface for providing RTP/RTCP contact.

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