[asterisk-users] asterisk 11.7.0: Delayed audio

[gm1]

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=???