Asterisk 11.7.0: Delayed Audio

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Asterisk Users 5 Comments

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 then not until about 6 seconds later I see this
>0xhexnumber — Probation passed – setting RTP source address to 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?


5 thoughts on - Asterisk 11.7.0: Delayed Audio

  • 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,

    What if any risk do i have with setting strictrtp=no with nat=no on a local network i.e.: 192.168.1.x ?


  • 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

  • I wanted to chime in on this one, I posted a similar problem a while back under the heading “Delay before audio starts” on 2/26/2013

    My solution to fix this problem was to adjust my dialplan by inserting an Answer();
    So I don’t think it necessarily has something to do with the strictrtp setting.