[Asterisk-Users] IAX2 audio problems but SIP OK?

[Steve Kann]

Whisker, Peter wrote:

>[sorry about previous mis-post]
>
>I have an * switch at home and one in the office. Both similar new CVS head
>versions and both with chan_sip2 built in:
>
>Asterisk CVS-HEAD-10/12/04-17:43:26
>Asterisk CVS-HEAD-10/13/04-12:53:52
>
>One is on a T1 connection and the other is on 576k/288k ADSL. The Ping time
>is about 30ms between the two servers, 90% of which is the ADSL delay.
>
>When I interconnect them with IAX2, I get rather choppy audio - with what
>sounds like dropped packets and jitter. 
>
>However when I interconnect with SIP is it clean and with no dropouts. The
>network path and timings are identical for both protocols and there is
>little noticeable difference when I play with the jitterbuffer setting in
>iax.conf.
>
>Does anyone have any idea why IAX protocol is causing this kind of problem?
>My ADSL is PPPoE which has an MTU of 1492 I think. Could this be causing the
>problems?
>  
>
No, you shouldn't be sending any IAX2 packets which approach this size.

1) What codecs are you using in each case?
2) Do you have other traffic on the link?  It's possible that somewhere 
along your path, the RTP audio traffic from SIP is getting some kind of 
helpful QoS benefit, while the IAX2 traffic is not?

-SteveK