[SIPForum-discussion] RTT estimation in SIP

[Sumeetkumar Bhardwaj]

Hi,

I don't think timer T1 reduces the retransmission number as it only control the time between two retransmission in UDP mode. The number of retransmission remain to 11 times regardless of timer T1 value for server transaction as per the RFC 3261. Only the time between two retransmission can be control using this.

Thanks,
Sumeet

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of mustafa rifaee
Sent: Friday, October 08, 2010 12:44 AM
To: Ruben Roque; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] RTT estimation in SIP

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<mailto: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<mailto: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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