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. -Gerard On Fri, 2014-01-10 at 15:01 -0600, 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 > > -- > 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 > -- Gerard Saraber <[email protected]> Rarcoa, Inc. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
