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[asterisk-users] strange SIP-SIP delay


 
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asterisk at dotr.com
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PostPosted: Tue Jun 17, 2008 10:39 am    Post subject: [asterisk-users] strange SIP-SIP delay Reply with quote

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 ?

Julian
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stotaro at totarotechn...
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PostPosted: Tue Jun 17, 2008 10:50 am    Post subject: [asterisk-users] strange SIP-SIP delay Reply with quote

On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith <asterisk at dotr.com> wrote:
Quote:
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 ?

Julian


Have you tested the latency across your VPN?

Thanks,
Steve T
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asterisk at dotr.com
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PostPosted: Tue Jun 17, 2008 11:08 am    Post subject: [asterisk-users] strange SIP-SIP delay Reply with quote

Hi Steve - the vpn is a "consistent" as the sip->IDSN has to go through
the VPN first to get to asterisk.

i.e. to make an "outside" call, PhoneA goes through the vpn to the
asterisk box, and out through isdn.

Julian

Steve Totaro wrote:
Quote:
On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith <asterisk at dotr.com> wrote:
Quote:
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 ?

Julian


Have you tested the latency across your VPN?

Thanks,
Steve T
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rj2807 at gmail.com
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PostPosted: Tue Jun 17, 2008 6:49 pm    Post subject: [asterisk-users] strange SIP-SIP delay Reply with quote

On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith <asterisk at dotr.com> wrote:
Quote:
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.
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