Martin Joseph wrote:
On 2006-10-25 08:14:43 -0700, "Noah Miller"
<[EMAIL PROTECTED]> said:
Hi Matt -
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
On the customer's end I have the following config in iax.conf:
trunk=no
(I have also tried trunk=yes and nothing for trunk=)
jitterbuffer=yes
forcejitterbuffer=yes
dropcount=3
minexcessbuffer=80
jittershrinkrate=1
If you're using Asterisk 1.2.x, dropcount, jittershrinkrate and
minexcesbuffer don't do anything. They are ignored by 1.2.x unless
you specify that you want to use the old 1.0.x jitterbuffer. Instead
you might try the parameters maxjitterbuffer, resyncthreshold, and
maxjitterinterps. For more, you can check out the sample iax.conf.
I believe, also, that you are correct in setting trunk=no. I know in
the 1.0.x jitterbuffer, trunk was not fully supported. I think this
is still the case with the 1.2.x jitterbuffer.
If the audio is dropping out completely, then I suspect the whole
jitter buffer thing is a red herring (waste of time).
Perhaps it's a nat issue? What kind of router if any is involved? I
am reaching here... Also, please do tell us which version of asterisk
you are running...
Marty
seeing this thread a lil too late, i guess. So, am sorry if I am
repeating things.
When I was setting up my iax2 configs, I too had one way audio initialy.
Tried the softphone on two machines(which incidentaly had asterisk
running on them as well), to no avail. When I looked at the tcpdump on
my asterisk server, I could see no rtp coming in from the two said machines.
So, I shifted the softphone to another machine, this time on a windows
machine, n voila! it worked like a charm.
So, I hope you did have a look at the tcpdump to check on the rtp flow.
cheerz
- Ben.
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