SIP Problem - Bad voice quality

We are experiencing a strange SIP problem.

We are running Asterisk 1.4 (we have tried different hardwares)

We have around 200 SIP phones registered (mostly Mitel, GXP 2000 and Cisco 7940)

For some reason, voice quality between SIP phones is awful. Echo, packet loss, etc…
Also, SIP trunks against other Asterisks running 1.4, 1.2.x and 1.0.x become awful, echo and packet loss.
Network is OK. No CRCs, collisions or errors in network. All of our ports are force to 100 FD.


Use reinvite between phones
Use IAX between Asterisks

But we still have some strange behavior, and Asterisk stops responding to SIP requests after a while. And we oftenly have crashes.

And… some SIP devices with whom we establish trunks work Ok with the Asterisk and do not experience the packet loss, echo, etc…
Also packet loss is present when SIP call originates from Asterisk 1.4 to Asterisk 1.0, but not vice-versa.

Any ideas ?
This sounds like a huge bug, we have been able to reproduce it with different hardwares, using different Asterisks to end the calls with SIP. If not a bug, what could it be ? Why would Asterisk crash oftenly, or stop responding to SIP requests ?



Everything started working back when we went back to Asterisk 1.2.15.
Just to keep everyone in touch of this problem, which someone else might also have.

if you want to help the 1.4 development along, it might be worth filing a bug report, although you’ll obviously need to supply some details/logs/dumps. you’ll probably also need to establish first whether it’s been fixed in trunk.

i’m experimenting with 1.4 on my home/test system, but i don’t think 1.4 is ready for production use just yet.

use * 1.2 instead 1.4