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
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org