Hi Fellows, We are working with our network team for audio/video traffic routing, some of the liks are not redundant and failover takes around 3 seconds to route the packets on the new link.
I'm trying to understand if we have some standard timers defined (have gone through the RFC 3550 and 4585) for RTP but couldn't find. My understanding is, transport for RTP is UDP, which does not offer reliability and left upto the application to determine the packet loss and inform the user. Are there any new developments around reliability and tolerance of RTP media for voice and video.Or any way to control the timers to handle the network level instability. Regards, Amanpreet Singh. _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
