.
Regards,
Aman
On Thu, Mar 28, 2024 at 11:56 PM Dale R. Worley wrote:
> Amanpreet Singh writes:
> > I'm looking to see if there is any RFC defining the standard around route
> > advance/ re-routing of calls to the next available route based on the SIP
> > failur
me call-id.
Seeking your help for RFC to refer to conclude this, thanks in advance.
Thanks,
Amanpreet Singh.
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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 wrote:
> Amanpreet Singh writes:
> > I'm looking for the best practices to have minimum packet loss/ delays
ssion 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 wrote:
> Amanpreet Singh writes:
> > We are wo
Arun, for what purpose would you like to inspect and differentiate the hold
and audio RTP packets?
and based on the signaling messages, can't that be achieved.
Thanks,
Amanpreet Singh.
On Thu, Sep 22, 2022 at 12:30 PM Arun Tagare wrote:
> Thanks Ranjit & Amanpreet, for your resp
Probably you can think of looking into the signaling messages(SDP in case
of SIP) to differentiate when the call is on hold and when not i.e. normal
audio RTP.
BTW what is the use case to differentiate call hold vs audio RTP?
Regards,
Amanpreet Singh.
On Wed, Sep 21, 2022 at 9:51 PM Arun
any way to control the timers to handle the
network level instability.
Regards,
Amanpreet Singh.
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