[asterisk-dev] Intermittent one-way audio and call failure on trunk

[Paul Belanger]

On 12-07-13 03:51 PM, Taylor, Jonn wrote:
>
> On 07/13/2012 08:15 AM, Matthew Jordan wrote:
>> ----- Original Message -----
>>> From: "Jonn Taylor" <jonnt at taylortelephone.com>
>>> To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
>>> Sent: Friday, July 13, 2012 7:47:30 AM
>>> Subject: Re: [asterisk-dev] Intermittent one-way audio and call
>>> failure on trunk
>>>
>>>
>>> On 07/12/2012 05:15 PM, Matthew Jordan wrote:
>>>> ----- Original Message -----
>>>>> From: "Jonn Taylor" <jonnt at taylortelephone.com>
>>>>> To: "Asterisk Developers" <asterisk-dev at lists.digium.com>
>>>>> Sent: Thursday, July 12, 2012 4:05:28 PM
>>>>> Subject: [asterisk-dev] Intermittent one-way audio and call
>>>>> failure on trunk
>>>>>
>> <snip>
>>
>>>>> IAX devices. SIP trunk provider is bandwidth.com(level3).
>>>> That's a lot of ground to cover. In particular, chan_unistim
>>>> received
>>>> major updates for Asterisk 11 (more information here:
>>>> https://wiki.asterisk.org/wiki/display/AST/Unistim+channel+improvements).
>>>>
>>>> You may want to try and narrow down the scope of your problems by
>>>> limiting things to one particular technology at a time.
>>> I have had one-way audio on SIP to SIP, SIP to IAX and SIP to USTM.
>> Can you provide the firmware and other specifications for the
>> actual devices you're having issues communicating with? When you have
>> issues communicating with them, is it consistent, or is it sporadic?
>>
>> <snip>
>>
>>>> 1) The first 200 OK we sent to the remote endpoint timed out. The
>>>> fact
>>>> that the endpoint failed to respond in a timely fashion, then
>>>> immediately
>>>> sent a BYE after ACK'ing the two 200 OKs (the original and the
>>>> re-transmit)
>>>> points towards some error in the remote end point - or at least, we
>>>> sent
>>>> it something it didn't like. This leads to...
>>>> 2) You're transmitting ICE candidates to the remote endpoint, which
>>>> may
>>>> be freaking it out. ICE is a new feature in Asterisk 11, and can
>>>> be
>>>> disabled in rtp.conf.
>>> Is this enabled by default?
>> Yes it is. Set 'icesupport' to false to disable it. If that does not
>> resolve your issues, we'll have to get some more information as to what
>> is actually happening. In that case, we'll re-open your JIRA issue and
>> track the debugging there.
>
> Disabling ICE helped a lot, just need to do more testing!!! Maybe the
> default should be disabled. I know that enabling new features encourages
> them to get used but this could cause a lot people trouble.
>
> Just my 2 cents.
>
I was just about to say something a long that lines too.  With an LTS 
release coming up, and having something as new as ICE enabled by default 
we might not be able to test everything properly.  It might be safer to 
default to disabled and have users make the choice to enable it.

-- 
Paul Belanger | PolyBeacon, Inc.
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