[asterisk-users] SIP audio delay after call transfer?

[Tony Mountifield]

I have a customer with an Asterisk 1.4 system (r144238 - between 1.4.22-rc5
and 1.4.22 released). It uses SIP to connect to the PSTN via a provider who
is on the same LAN as the box (it is co-located at the provider). They also
have about 20 SIP phones as extensions that connect to the box over the
internet. "sip show peers" indicates that most phones have a latency of
90ms-100ms. The provider is at 1ms. All links use the digium G.729 codec.

They have reported that while call quality is normally very good, if a call
is transferred from one extension to another, the transferred call starts
to experience considerable audio latency. Transferring the call again also
increases this latency even more, such that the call is unusable.

My suspicion is that while performing the transfer, audio frames are building
up somewhere and not being flushed (lack of autoservice somewhere in the code?).

Has anyone else observed this behaviour? Even better, has anyone got a fix,
or knows of such an issue having been fixed in a later version?

This is a production system, so I can't easily try different versions to
experiment, but could justify the downtime to install a known solution.

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org