Thanks Ranjit, James and Dale for your valuable inputs.
I think now I have some way forward.

Regards,
Amanpreet Singh.


On Mon, Sep 26, 2022 at 5:31 AM Dale R. Worley <wor...@ariadne.com> wrote:

> Amanpreet Singh <amanpreeet.si...@gmail.com> writes:
> > 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.
>
> If you're sending real-time data, such as a two-way conversation, if the
> link loses 3 seconds of data, you're going to lose 3 seconds of data.
> Video and audio codecs use various tricks to cover up single packet
> losses by synthesizing data to fill the gap in a way that avoids humans
> perceiving the gap.  But there's no way to do that with a 3 second gap,
> the recipient will notice.
>
> If you're sending one-way data in near-real-time, and the major losses
> are three-second gaps, you could send the audio data twice, with a
> three-second offset between them.  A suitable decoder can patch the two
> together to cover 3-second losses.  RFC 2198 gives a framework for
> sending RTP packets that contain both a primary stream and a delayed,
> redundant secondary stream.  But unfortunately, RFC 2198 specifies a
> 14-bit field to give the time offset between the streams, and with a
> typical 8k/sec sample rate, the maximum value that can be specified is
> 2.048 seconds, which isn't long enough for your situation.
>
> However, you could specify two separate RTP streams in the SDP, one of
> which is the same packets as the other but transmitted 3 or more seconds
> later.
>
> Dale
>
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