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

[Pavel Jezek]

if anyone has one-way audio issues with iax over jittery connection, 
please look at bug report, what I created yesterday and report your 
experiences,
I think this is one of the most serious bug, that must be identified and 
resolved before 1.4 will be released, thanks
http://bugs.digium.com/view.php?id=8325
PJ






Benjamin Jacob wrote:
> 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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