Re: [asterisk-users] RTP keepalive doesn't work
Kevin P. Fleming kpfleming at digium.com writes: Yes, it was lost during a merge of code into Asterisk trunk after 1.6.2 was branched (so only 1.8.0 and trunk are missing the code). Leif Madsen entered an issue on Mantis as a blocker for any more 1.8.x releases until this is resolved, as it is clearly a regression in the 1.8.x series. So is this working in which release 1.6.2.18 ? -- _ -- 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] RTP keepalive doesn't work
Hi Kevin, Did you have any luck tracking down the missing rtpkeepalive code? I'm really looking to get this working asap so I'd be happy to copy in/compile/trial some code if there's any available. Regards, Ryan. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Saturday, 29 January 2011 1:33 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] RTP keepalive doesn't work On 01/28/2011 09:24 AM, Ryan Tucker wrote: Thanks for the info, I guess I would expect asterisk to send 'silence' (in blank RTP form or something) if silence suppression is disabled. Just as I would expect any end point to send 'silence' if it was muted when silence suppression was disabled. It seems that RTP keepalives would serve this purpose, however this doesn't seem to be available either... Should I file a bug report re rtpkeepalive? No need... I'm already trying to track down when the code was removed, and for what reason. Once that is done I'll enter an issue to get it addressed. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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 -- _ -- 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] RTP keepalive doesn't work
On 02/04/2011 01:34 AM, Ryan Tucker wrote: Did you have any luck tracking down the missing rtpkeepalive code? I'm really looking to get this working asap so I'd be happy to copy in/compile/trial some code if there's any available. Yes, it was lost during a merge of code into Asterisk trunk after 1.6.2 was branched (so only 1.8.0 and trunk are missing the code). Leif Madsen entered an issue on Mantis as a blocker for any more 1.8.x releases until this is resolved, as it is clearly a regression in the 1.8.x series. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] RTP keepalive doesn't work
On 01/27/2011 10:52 PM, Ryan Tucker wrote: So, I've done some more testing and got some more info. I have one endpoint that does silence suppression and one that doesn't. When the silence suppressing endpoint stops sending RTP, asterisk stops sending RTP to the other endpoint. I have disabled directmedia and directrtpsetup and it made no difference. I have even forced one endpoint to use GSM and the other to use ULAW (forcing asterisk to re encode everything) and asterisk STILL stops sending RTP when the endpoint does... Asterisk doesn't have anything to send. What do you expect it to send when it's not receiving anything? I see that we have an rtpkeepalive configuration option, but I don't see that any code actually causes keepalive packets to be sent anywhere... it did when it was first added, but somehow that code has been lost. This certainly warrants some investigation to find out when it was removed and why, because the configuration option should have been removed if the keepalive support was removed on purpose. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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] RTP keepalive doesn't work
Thanks for the info, I guess I would expect asterisk to send 'silence' (in blank RTP form or something) if silence suppression is disabled. Just as I would expect any end point to send 'silence' if it was muted when silence suppression was disabled. It seems that RTP keepalives would serve this purpose, however this doesn't seem to be available either... Should I file a bug report re rtpkeepalive? Sent from my iPhone On 29/01/2011, at 12:55 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 01/27/2011 10:52 PM, Ryan Tucker wrote: So, I've done some more testing and got some more info. I have one endpoint that does silence suppression and one that doesn't. When the silence suppressing endpoint stops sending RTP, asterisk stops sending RTP to the other endpoint. I have disabled directmedia and directrtpsetup and it made no difference. I have even forced one endpoint to use GSM and the other to use ULAW (forcing asterisk to re encode everything) and asterisk STILL stops sending RTP when the endpoint does... Asterisk doesn't have anything to send. What do you expect it to send when it's not receiving anything? I see that we have an rtpkeepalive configuration option, but I don't see that any code actually causes keepalive packets to be sent anywhere... it did when it was first added, but somehow that code has been lost. This certainly warrants some investigation to find out when it was removed and why, because the configuration option should have been removed if the keepalive support was removed on purpose. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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 -- _ -- 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] RTP keepalive doesn't work
On 01/28/2011 09:24 AM, Ryan Tucker wrote: Thanks for the info, I guess I would expect asterisk to send 'silence' (in blank RTP form or something) if silence suppression is disabled. Just as I would expect any end point to send 'silence' if it was muted when silence suppression was disabled. It seems that RTP keepalives would serve this purpose, however this doesn't seem to be available either... Should I file a bug report re rtpkeepalive? No need... I'm already trying to track down when the code was removed, and for what reason. Once that is done I'll enter an issue to get it addressed. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.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
[asterisk-users] RTP keepalive doesn't work
Hey guys, I'm using asterisk 1.6.2.13 and have an endpoint which uses silence suppression which I can't turn off. I've set rtpkeepalive=10 in sip.conf [general], as well as under the peer details for our sip provider but it doesn't seem to do anything. Rtp debug shows that we are receiving RTP from the SIP provider, and forwarding it to the end point, but no RTP packets are sent back to the provider (ie. No keep alives). I did find a bug report of this exact issue, but it was closed with the message to ask the mailing list... Any ideas? -- _ -- 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] RTP keepalive doesn't work
So, I've done some more testing and got some more info. I have one endpoint that does silence suppression and one that doesn't. When the silence suppressing endpoint stops sending RTP, asterisk stops sending RTP to the other endpoint. I have disabled directmedia and directrtpsetup and it made no difference. I have even forced one endpoint to use GSM and the other to use ULAW (forcing asterisk to re encode everything) and asterisk STILL stops sending RTP when the endpoint does... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Tucker Sent: Friday, 28 January 2011 11:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion (asterisk-users@lists.digium.com)' Subject: [asterisk-users] RTP keepalive doesn't work Hey guys, I'm using asterisk 1.6.2.13 and have an endpoint which uses silence suppression which I can't turn off. I've set rtpkeepalive=10 in sip.conf [general], as well as under the peer details for our sip provider but it doesn't seem to do anything. Rtp debug shows that we are receiving RTP from the SIP provider, and forwarding it to the end point, but no RTP packets are sent back to the provider (ie. No keep alives). I did find a bug report of this exact issue, but it was closed with the message to ask the mailing list... Any ideas? -- _ -- 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 -- _ -- 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] Rtp keepalive
Hi, I've got a problem with rtp keepalives. I'm using basically the same config on 2 hosts, but one of them sends rtp comfort noise when it's on hold, the other doesn't. The only difference I can think of now is that one of the machines is multihomed, but that might be unrelated. rtpkeepalive is set to 2 and I can confirm is by doing `sip show settings`. I've tried all combinations of nat and qualify for the peer that has problems - rtp comfort noise is simply not sent. After trying to make it work for a day or so, I reported it as a bug (https://issues.asterisk.org/view.php?id=15466) but maybe someone here has some ideas how to make it work? -- Kind regards, Stanisław Pitucha, Gradwell Voip Engineer T: 01225 800 851 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com Gradwell - Internet for Business People Phone Services | Business Broadband | Email Website Hosting Can switching to VoIP today put some change in your pocket? Registered Address: 26 Cheltenham Street, Bath, BA2 3EX, UK. Company Number: 3673235 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
