[asterisk-users] Re: IAX2 goes "one way audio" when lag gets bad

Benjamin Jacob benjamin.jacob at masconit.com
Wed Oct 25 20:53:02 MST 2006


Martin Joseph wrote:

> On 2006-10-25 08:14:43 -0700, "Noah Miller" 
> <noahisaacmiller at gmail.com> 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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