[asterisk-users] asterisk 11.7.0: Delayed audio

[gm1]

On 01/13/2014 10:09 AM, gm1 wrote:
> On 01/10/2014 08:33 PM, gm1 wrote:
>> On 01/10/2014 04:01 PM, Matthew Jordan wrote:
>>> On Fri, Jan 10, 2014 at 9:45 AM, gm1 <gm1 at curtissystemssoftware.com> 
>>> wrote:
>>>> On connection to an incoming call via PSTN where
>>>> asterisk [11.7.0] is Dialing an internal extension
>>>> on answering the call there is about 6-7 seconds before
>>>> audio is heard on either side.
>>>>
>>>>
>>>> When looking at the CLI traces when I answer the incoming call that 
>>>> asterisk
>>>> extensions were dialing, I see immediately upon answering
>>>>> 0xhexnumber -- Probation passed - setting RTP source address to
>>>>> 192.168.1.11:portnumber
>>>> then not until about 6 seconds later I see this
>>>>> 0xhexnumber -- Probation passed - setting RTP source address to
>>>>> 192.168.1.11:diffportnumber
>>>> and immediately hear audio
>>>>
>>>> what appears to be an issue is that the RTP link(audio) setup is 
>>>> delayed.
>>>>
>>>>
>>>> Anyone have suggestions on how to fix this issue?
>>>>
>>> If the RTP source address/port is changing, then Asterisk is receiving
>>> RTP packets from two different sources and is waiting for one of them
>>> to stabilize before it picks the actual source of the media stream.
>>> That's by design, as the "locking in" of the RTP source prevents a
>>> media injection attack.
>>>
>>> You can tweak how Asterisk does this using two settings in rtp.conf:
>>>
>>> ; Enable strict RTP protection. This will drop RTP packets that
>>> ; do not come from the source of the RTP stream. This option is
>>> ; enabled by default.
>>> ; strictrtp=yes
>>>
>>> ; Number of packets containing consecutive sequence values needed
>>> ; to change the RTP source socket address. This option only comes
>>> ; into play while using strictrtp=yes. Consider changing this value
>>> ; if rtp packets are dropped from one or both ends after a call is
>>> ; connected. This option is set to 4 by default.
>>> ; probation=8
>>>
>>> Matt
>>>
>> Matt,
>>
>> What if any risk do i have with setting strictrtp=no
>> with nat=no on a local network i.e.: 192.168.1.x  ?
>>
>> pc
>>
> I changed strictrtp=no and restarted asterisk,
> no difference in delayed audio ... still near 6 seconds.
> In cli when I answer the incoming call I see asterisk immediately show 
> answer.
>
> Perhaps this issue is caused by something other than the strictrtp 
> setting?
>
> what are all the possible settings for strictrtp=???
>
we have yet no resolution ...
Does any one have any suggestions where to place some printf s to 
understand after a call is answered
what is delaying the audio ?  I am building source 11.7.0