RTP is usually used to carry audio and video media. The analysis of how to manage error recovery will likely begin with understanding the nature of the RTP media traffic. If it is two-way real time conversation (e.g., Zoom, Teams, etc.) then there is generally not a good way to retransmit anything lost. In this case, the best you can do is minimize the delay to establish the alternate routes. However, if the media is one-way streaming (e.g., Netflix), then a scheme that includes resending X milli-seconds of lost media would likely provide a better customer experience if the application is buffered appropriately. ... James Bress
-----Original Message----- From: sip-implementors-boun...@lists.cs.columbia.edu <sip-implementors-boun...@lists.cs.columbia.edu> On Behalf Of Amanpreet Singh Sent: Sunday, September 25, 2022 9:10 AM To: Dale R. Worley <wor...@ariadne.com> Cc: discuss...@sipforum.org; sip-implementors <sip-implementors@lists.cs.columbia.edu> Subject: Re: [Sip-implementors] [SIPForum-discussion] RTP packet loss tolerance/ timers Dale, I'm looking for the best practices to have minimum packet loss/ delays in case of primary network link failure. As the network secondary link takes about 3 seconds to come up. What best we can do on the application side to have the minimum RTP packet loss? Do we have timeout, retransmission timers for RTP, or any mechanism to adjust to minimize the loss. if not adjustable, default values based on which we can try changing network layer failover. Thanks, Amanpreet Singh. On Sun, Sep 25, 2022 at 7:38 AM Dale R. Worley <wor...@ariadne.com> wrote: > 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 _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
