On 01/10/2014 04:01 PM, Matthew Jordan wrote:
On Fri, Jan 10, 2014 at 9:45 AM, gm1 <[email protected]> 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
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