[SIPForum-discussion] RTT estimation in SIP

[mustafa rifaee]

Hello Mr.Ruben;
Thanks for your reply;
I am doing research in VOIP Signaling, and i made a comparison analysis for
using SIP over UDP/TLS/TCP, and we know that the UDP protocol give us the
best performance, but the UDP  problem is it unreliable protocol and SIP use
it's timers with UDP, but with bad conditions network and with small value
for timer A (RTT), it will be storm of SIP request retransmission.
If we can estimate a adaptive value for  RTT according to these network
condition we will reduce the SIP request retransmission number.

if we can estimate RTT using Time Stamp header in SIP, Is this useful ?

Thanks
Mustafa

On Thu, Oct 7, 2010 at 2:43 PM, Ruben Roque <decruzreuben at gmail.com> wrote:

> Hi Mustafa,
>
> I would like you to read RFC 3261 section 17 clearly once again.
>
> Try and understand why the timer A is required, would happen without
> timer A? And i think that the RTT is mandatoraly fixed but at the same
> time it is variable based on a particular condition - try to
> understand what?
>
> Read and let me know.I can make you understand more in depth later.
>
> Secondly, do you know why the INVITE request takes some time to
> generate a final response?
>
>
> Regards,
> Reuben
>
> On Wed, Oct 6, 2010 at 10:15 PM, mustafa rifaee
> <mustafa.rifaee at gmail.com> wrote:
> >
> > Hi all;
> > I would like to ask you if there is any important in SIP to find a
> mechanism
> > to estimate and calculate RTT (T1 timer).
> > for example in RFC 3261 we can estimate RTT using provisional response
> 100
> > Trying and TimeStamp for INVITE Transaction.
> > and if choice an error value for RTT  ( T1 Timer), is this will effect in
> > Call setup time Delay?
> >
> > Please Help me;
> > Thanks
> > Mustafa
> >
> >
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