[Asterisk-Users] Hickup and missing voice

[Rich Adamson]

> Let me run this by the group, my inter-IAX connections are extremely 
> pop-hickup,missing syllable type of affairs.
> Played with jitter buffers, card levels (this is totally independent of any 
> level setting) and about anything else we can adjust. Did the adjustments, 
> undid them. Doesn't matter, problem seems to be uniform. Level doesn't matter.
> Ive got a 4 mb cable connection that looks extremely robust between the two 
> penguins (Mandrake 9.2) and it still occurs. Running Digium TDM400P cards 
> on both boxes, (which I might add are absolutely BEAUTIFULLY made boards) 
> both are mirrors of each other. We have no problems talking, just the talk 
> is choppy...bits and pieces choke...Ive had 3-5 seconds of blank,silent air 
> on some calls when I know the other end was talking.
> Im sure Im missing something extremely simple here. As a person with a 
> bunch of PSTN experience, I'd think it was a timing problem, tearing up the 
> codecs randomly, I'm using GSM. Any input here might be helpful...as I'm 
> about at wits end. Ive run diags over the link, and theres NO dropped 
> packets, or other crapoola..
> Slap me with a baseball bat, Ive looked at the wiki, the beta docs, 
> hitchhikers guide, the swiss-army-knife docs...all the stuff..Im beating my 
> head against the curb. Im sure Im missing something someone will say is so 
> simple it's a DUH! when I find out.
> PSTN likes to echo a lot, Versleazon most notably, when someone from their 
> PSTN joins a conf, conferences loaded to the max or only two phones are 
> equally as bad. Im wearing a bald spot scratching my head...help?

Several thoughts come to mind, but since you didn't include any comments
as to what type of phones you're using, etc, these are simply guesses.

1. As someone else mentioned, if you're using an older version of code,
you might first try upgrading.

2. If you're using Cisco phones, upgrade to latest dev (head) cvs. There
has been issues with iax/gsm transcoding to sip/rtp that have been slowly
getting fixed. (Part of those problems relate to how cisco deals with
uneven timestamps within the rtp stream.)

3. Several phones provide an option that essentially means "transmit silence".
The default for the option is frequently "transmit silence = no" which will
cause choppy sound. (In Xlite for exampe, its under Advanced Settings.)
Change the option to yes.

4. If you have other hubs/switches/infrastructure components involved in
the end-to-end path, might consider checking those to ensure packets are
being dropped, collisions are reasonable, etc. One way to check is to use
ethereal (or some other packet sniffer) to observe a real conversation,
and look at the flow of packets to ensure they are received/transmitted
in an even and consistent stream.

Rich