[Asterisk-Users] IAX voice distortion with full upload channel / SIP ok

[Rich Adamson]

> this is the scenario:
> 
> One * is placed in a central location with more than enough up/down
> bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via
> IAX trunking. Everything is fine until the upload channel of the remote
> site is filled with a download, then heavy voice distortion starts. Well
> of course this is expected. So I fooled around with HFSC QoS scheduling
> on the remote site Linux machine. The scheduling seems to work as the
> TOS marked traffic is put in the correct queue and the upload bandwith
> for other applications is going down.
> 
> BUT: The voice quality problems definatly stay when using IAX. The funny
> part: Doing the same with SIP shows no big problems. SIP calls to
> T-Online work nicely and even if I change the * <-> * link from IAX to
> SIP everything is fine even with full up-/downloads on the remote DSL
> connection.
> 
> My conclusion would be that this depends on the IAX implementation
> somehow. I tried different settings for jitterbuffer, trunk and
> trunktimestamp all with the same result. This currently means we go back
> to SIP for *<->* linkage.
> 
> Any ideas? If I should rather post this over in -dev please let me know!

The iax problems tend to be oriented around version issues. Many of the
itsp's have added whatever functionality they needed to asterisk to 
support their operation, and upgrading their code to the latest levels
is not a trevial task.

Given the changes that have occurred in the iax code over the last
year or so, mismatches in iax versions are known to cause significant
audio quality issues. Turning off the jitterbuffer, trunk=no, etc, is
oftentimes the only way to get close to reasonable audio quality.

Also, you might experiment with different codecs over iax links as you'll 
find some that are better then others. That should _not_ be interpreted 
as the lowest bandwidth codec is the best in every case.

Last, since most of the itsp's won't tell you what versions they are
running, expect changes over time. E.g., they might upgrade their code
to a later version without any notification and suddently your call
quality changes for no apparent reason. (The same is generally not an
issue with sip-based interfaces as there have been far fewer changes
to date in it that impact call quality.)