Re: [asterisk-users] changing the size of voice packets
John Todd wrote: There was discussion recently (on -dev? on -users? on IRC?) about how there are some shortcomings on RTP packetization/transcoding. It appears, though I have not confirmed this, that trying to move a 20ms G.711 stream from a client, though Asterisk, to a remote gateway using 40ms G.711 will NOT work correctly. The 20ms packet size is passed through without aggregating to 40ms, or vice versa - no change in packetization (though I don't know which side takes precedence.) Going the opposite directon for dis-aggregation (which is what you want to do) I assume would fail in similar ways. I don't recall if changing the codec made any difference on the packetization between two bridged channels. In the past (trunk pre-1.4 and 1.4) both handled aggregation properly, with one important caveat: 1. The media actually flows through Asterisk (no RTP re-invites) If the media is re-invited, it is up to the clients/peers To honor the packetization the remote end requested. If the media is not reinvited and is 100% compatible, codec and packetization, it will go through the packet-to-packet bridge. At one point the P2P bridge did not know about packetization differences and would just relay the RTP packets. I believe that was fixed a long time ago. For what it's worth, 10ms is the maximum rate for most codecs. This creates twice as many packets as 20ms, three times as many as 30ms, etc. - hopefully your network hardware has sufficient power or your call volumes are reasonably low so as not to produce an overwhelming number of Packets Per Second (PPS). Decreasing sampling interval also gets you closer to reaching your NIC's threshhold of PPS, which often is not huge. I seem to recall asking the person who reported that to open a bug in Mantis, but I can't find it, though I didn't look exhaustively. If you can verify this and/or it's relevant to you, please open a ticket so that it at least will be reviewed. I'd open it myself, but I'm a bit resource constrained at the moment in an airport lobby. ___ -- 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
Re: [asterisk-users] changing the size of voice packets
is any command , shows the current rate of each channel? --- On Mon, 11/10/08, Kristian Kielhofner [EMAIL PROTECTED] wrote: From: Kristian Kielhofner [EMAIL PROTECTED] Subject: Re: [asterisk-users] changing the size of voice packets To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, November 10, 2008, 11:10 PM On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov [EMAIL PROTECTED] wrote: If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? Depends? What is the status of maxptime in Asterisk? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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 ___ -- 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
Re: [asterisk-users] changing the size of voice packets
--- On Mon, 11/10/08, Igor Goncharovsky [EMAIL PROTECTED] wrote: From: Igor Goncharovsky [EMAIL PROTECTED] Subject: Re: [asterisk-users] changing the size of voice packets To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Monday, November 10, 2008, 3:00 PM Hi! On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali [EMAIL PROTECTED] wrote: Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. You can specify size of voice packets in allow line of sip.conf peer configuration. ex.: allow=alaw:30,g729:50 thanks for your great help 2 questions 1- in allow=g729:50 50 is time or size of packet for smaller packet, 40 or 60 ? 2- what about for asterisk 1.2 For more information: http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+rtp-packetization.txt -- Best regards, Igor Goncharovsky ___ -- 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
[asterisk-users] changing the size of voice packets
Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. thanks in advance Mani ___ -- 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
Re: [asterisk-users] changing the size of voice packets
Hi! On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali [EMAIL PROTECTED]wrote: Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. You can specify size of voice packets in allow line of sip.conf peer configuration. ex.: allow=alaw:30,g729:50 For more information: http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+rtp-packetization.txt -- Best regards, Igor Goncharovsky ___ -- 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
Re: [asterisk-users] changing the size of voice packets
Igor Goncharovsky wrote: You can specify size of voice packets in allow line of sip.conf peer configuration. ex.: allow=alaw:30,g729:50 If you are interfacing with any commercial VoIP equipment/media gateway equipment, do be aware that packetisation durations for G.711u and G.729 that are not 10, 20, or 30 ms are not widely supported, and it may be that 20 ms is the only duration supported -- the most common. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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
Re: [asterisk-users] changing the size of voice packets
On Nov 10, 2008, at 3:30 AM, Igor Goncharovsky wrote: Hi! On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali [EMAIL PROTECTED] wrote: Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. You can specify size of voice packets in allow line of sip.conf peer configuration. ex.: allow=alaw:30,g729:50 For more information: http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+rtp-packetization.txt -- Best regards, Igor Goncharovsky There was discussion recently (on -dev? on -users? on IRC?) about how there are some shortcomings on RTP packetization/transcoding. It appears, though I have not confirmed this, that trying to move a 20ms G.711 stream from a client, though Asterisk, to a remote gateway using 40ms G.711 will NOT work correctly. The 20ms packet size is passed through without aggregating to 40ms, or vice versa - no change in packetization (though I don't know which side takes precedence.) Going the opposite directon for dis-aggregation (which is what you want to do) I assume would fail in similar ways. I don't recall if changing the codec made any difference on the packetization between two bridged channels. For what it's worth, 10ms is the maximum rate for most codecs. This creates twice as many packets as 20ms, three times as many as 30ms, etc. - hopefully your network hardware has sufficient power or your call volumes are reasonably low so as not to produce an overwhelming number of Packets Per Second (PPS). Decreasing sampling interval also gets you closer to reaching your NIC's threshhold of PPS, which often is not huge. I seem to recall asking the person who reported that to open a bug in Mantis, but I can't find it, though I didn't look exhaustively. If you can verify this and/or it's relevant to you, please open a ticket so that it at least will be reviewed. I'd open it myself, but I'm a bit resource constrained at the moment in an airport lobby. JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- 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
Re: [asterisk-users] changing the size of voice packets
If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? On Mon, November 10, 2008 2:17 pm, John Todd wrote: On Nov 10, 2008, at 3:30 AM, Igor Goncharovsky wrote: Hi! On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali [EMAIL PROTECTED] wrote: Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. You can specify size of voice packets in allow line of sip.conf peer configuration. ex.: allow=alaw:30,g729:50 For more information: http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+rtp-packetization.txt -- Best regards, Igor Goncharovsky There was discussion recently (on -dev? on -users? on IRC?) about how there are some shortcomings on RTP packetization/transcoding. It appears, though I have not confirmed this, that trying to move a 20ms G.711 stream from a client, though Asterisk, to a remote gateway using 40ms G.711 will NOT work correctly. The 20ms packet size is passed through without aggregating to 40ms, or vice versa - no change in packetization (though I don't know which side takes precedence.) Going the opposite directon for dis-aggregation (which is what you want to do) I assume would fail in similar ways. I don't recall if changing the codec made any difference on the packetization between two bridged channels. For what it's worth, 10ms is the maximum rate for most codecs. This creates twice as many packets as 20ms, three times as many as 30ms, etc. - hopefully your network hardware has sufficient power or your call volumes are reasonably low so as not to produce an overwhelming number of Packets Per Second (PPS). Decreasing sampling interval also gets you closer to reaching your NIC's threshhold of PPS, which often is not huge. I seem to recall asking the person who reported that to open a bug in Mantis, but I can't find it, though I didn't look exhaustively. If you can verify this and/or it's relevant to you, please open a ticket so that it at least will be reviewed. I'd open it myself, but I'm a bit resource constrained at the moment in an airport lobby. JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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
Re: [asterisk-users] changing the size of voice packets
On Mon, Nov 10, 2008 at 2:40 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: Depends? What is the status of maxptime in Asterisk? ... or the remote end, for that matter... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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
Re: [asterisk-users] changing the size of voice packets
On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov [EMAIL PROTECTED] wrote: If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? Depends? What is the status of maxptime in Asterisk? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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
Re: [asterisk-users] changing the size of voice packets
Well, I should say irreconcilably different, not different. On Mon, November 10, 2008 2:40 pm, Kristian Kielhofner wrote: On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov [EMAIL PROTECTED] wrote: If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? Depends? What is the status of maxptime in Asterisk? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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
Re: [asterisk-users] changing the size of voice packets
Alex Balashov wrote: If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? Only if the 'ptime' or 'maxptime' values offered are not legal for the codec involved; if they are supported by the codec, then 'ptime' is just a preference, not a demand. The endpoint accepting the offer is free to send packets of any legal size, but not exceeding 'maxptime'. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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
Re: [asterisk-users] changing the size of voice packets
Kevin P. Fleming wrote: Alex Balashov wrote: If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? Only if the 'ptime' or 'maxptime' values offered are not legal for the codec involved; if they are supported by the codec, then 'ptime' is just a preference, not a demand. The endpoint accepting the offer is free to send packets of any legal size, but not exceeding 'maxptime'. Hm. Then perhaps the instances in which I've seen this play out have been conflicting maxptime values. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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
Re: [asterisk-users] changing the size of voice packets
On Mon, Nov 10, 2008 at 4:08 PM, Kevin P. Fleming [EMAIL PROTECTED] wrote: Alex Balashov wrote: If the packetisation durations are different between endpoints, the SDP offer/answer should fail with a 488 Not Acceptable Here. Right? Only if the 'ptime' or 'maxptime' values offered are not legal for the codec involved; if they are supported by the codec, then 'ptime' is just a preference, not a demand. The endpoint accepting the offer is free to send packets of any legal size, but not exceeding 'maxptime'. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) Yep, that's how it's supposed to work. Are you confirming our understanding of the spec or Asterisk's implementation of the spec? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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
Re: [asterisk-users] changing the size of voice packets
Kristian Kielhofner wrote: Are you confirming our understanding of the spec or Asterisk's implementation of the spec? Well the former, and I hope the latter too, since it should match the former :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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
Re: [asterisk-users] changing the size of voice packets
On Mon, Nov 10, 2008 at 4:52 PM, Kevin P. Fleming [EMAIL PROTECTED] wrote: Kristian Kielhofner wrote: Are you confirming our understanding of the spec or Asterisk's implementation of the spec? Well the former, and I hope the latter too, since it should match the former :-) -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) Excellent answer! :) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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
