[asterisk-users] asterisk 11.7.0: Delayed audio

[gm1]

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