Amanpreet Singh <amanpreeet.si...@gmail.com> writes: > 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.
I haven't heard of any. But could you provide some detail what the usage would be of a timer that you're looking for? Dale _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
