[asterisk-users] strange SIP-SIP delay

[Raj Jain]

On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith <asterisk at dotr.com> wrote:
> I've got the following setup:
>
> PhoneA ->
>  router ->
>   vpn ->
>    router->
>     asterisk (SIP / ISDN)
>
> PhoneB ->
>  asterisk (SIP / ISDN)
>
> If phone A is connected to phone B (sip-sip), there is a noticable delay
> (up to 2-3 seconds) between me speaking and the other end hearing.
>
> If phone A calls out via the ISDN and back in  to the DDI of phone B (ie
> SIP->ISDN->ISDN->SIP) then there is no delay at all !
>
> Any clues on where I might start looking for this ?
>

Are you using canreinvite=yes setting (i.e. is the RTP media expected
to flow directly between the phones as opposed to hair-pining through
Asterisk)? If so, some of the delay could be attributed to the time
spent in RE-INVITEs that happen after the call set up.

--
Raj Jain

P.S. There is the directrtpsetup= flag that can eliminate this latency
(if you're indeed using canreinvite=yes), but I believe that feature
is considered "experimental". Actually, if that feature is still
experimental, I'd like to change that and fix any associated bugs
because it seems like a pretty useful feature to me for people who
want to use Asterisk as a call controller (a.k.a. soft-switch) that
does not need to participate in the media path.