Re: [asterisk-users] asterisk 11.7.0: Delayed audio
On 01/13/2014 10:09 AM, gm1 wrote: On 01/10/2014 08:33 PM, gm1 wrote: On 01/10/2014 04:01 PM, Matthew Jordan wrote: On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@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 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 -- _ -- 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
Re: [asterisk-users] asterisk 11.7.0: Delayed audio
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 g...@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 -- Gerard Saraber gsara...@rarcoa.com 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
Re: [asterisk-users] asterisk 11.7.0: Delayed audio
On 01/10/2014 08:33 PM, gm1 wrote: On 01/10/2014 04:01 PM, Matthew Jordan wrote: On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@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 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=??? -- _ -- 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
[asterisk-users] asterisk 11.7.0: Delayed audio
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? pc -- _ -- 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
Re: [asterisk-users] asterisk 11.7.0: Delayed audio
On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@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 -- _ -- 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
Re: [asterisk-users] asterisk 11.7.0: Delayed audio
On 01/10/2014 04:01 PM, Matthew Jordan wrote: On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@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 -- _ -- 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
