Re: [asterisk-users] Improving the voice Quality,
Steve, Steve Totaro wrote: BT3 (BackTrack) LiveCD is one of the best things out there, even has sipp built right in, as well as other great apps, utilities, and security auditing. I suggest everyone have a copy in their arsenal, and it is free of course. What does it do? I mean, for someone like you, practically speaking? -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Improving the voice Quality,
All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? Any thoughts? -Thank you, -Jai ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. -- 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delaytos=0x08 high throughput tos=0x04 high reliabilitytos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
On Fri, 3 Oct 2008, Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. Not quite RTP traffic, but iperf might help - if you have access to both ends of the link. You can set a bandwidth and packet size which might help you. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
Oh yes, how could I forgot about that? Thank you, -Jai On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov [EMAIL PROTECTED]wrote: sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
All, Just an update on this. This turned out to be a bug in Cisco firewall. We ended up in upgrading the Firmware on the firewall. One thing I want to add, this was first time we used the fail over unit during peak time. In the whole process (failover, upgrade and failover back to active unit) was completely seamless. Did not had any down time, there was just a pause for just 1 second in the audio. I was very impressed. -Jai On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi [EMAIL PROTECTED] wrote: Oh yes, how could I forgot about that? Thank you, -Jai On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov [EMAIL PROTECTED]wrote: sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving the voice Quality,
BT3 (BackTrack) LiveCD is one of the best things out there, even has sipp built right in, as well as other great apps, utilities, and security auditing. I suggest everyone have a copy in their arsenal, and it is free of course. Thanks, Steve Totaro On Fri, Oct 3, 2008 at 10:09 PM, Jai Rangi [EMAIL PROTECTED] wrote: All, Just an update on this. This turned out to be a bug in Cisco firewall. We ended up in upgrading the Firmware on the firewall. One thing I want to add, this was first time we used the fail over unit during peak time. In the whole process (failover, upgrade and failover back to active unit) was completely seamless. Did not had any down time, there was just a pause for just 1 second in the audio. I was very impressed. -Jai On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi [EMAIL PROTECTED] wrote: Oh yes, how could I forgot about that? Thank you, -Jai On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov [EMAIL PROTECTED]wrote: sipp can simulate RTP traffic. Jai Rangi wrote: Al and Alex, Thank you for your input, Sorry TDM is not the option at this time :( . Everything has been great until last 2-3 days. Machine loads is not the issue, we have multiple asterisk server to share the load. Not much change in traffic. Now it been narrowed down to networking and we are trying to find out where the issue is? In our Firewall or SP's router. Does any one know of any tool to simulate RTP traffic. Its pain to find out the bad calls out of hundreds of calls. BTW, What should be right value for tos in sip.conf. We have tos=0x68 Dont remember how did I come up with this value. I found this http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos tos=0x10 low delay tos=0x08 high throughput tos=0x04 high reliability tos=0x02 ECT bit set tos=0x01 CE bit set -Jai On Fri, Oct 3, 2008 at 4:58 AM, Al Baker [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: USE TDM Circuits - Voice Quality Good Alex Balashov wrote: Jai Rangi wrote: All, I am having audio quality problem in some calls (1-2%) on asterisk. I captured RTP traffic using ethereal and this is what I found with the problematic calls. (Worst cases) Drop by Jitter buff: 25-75% Out of Seq: 50-100% (100 % means very very poor call quality). Has anyone had similar problem? If yes, can you please share your experience on how did you fix this? Such poor performance is not fixable. The network, connectivity issues, machine load, etc. needs to be addressed - the underlying cause, in other words. BTW, 100% out-of-sequence RTP packets leads to a lot more than just very very poor call quality. I don't see how the conversation could even be coherent in that situation. What is more likely is that Wireshark's RTP stats are giving you some distorted information. I've found its stream analysis to be somewhat buggy in that regard. I was wondering if I can decrease the MTU size to 250-500 on the network card and use that card only for VoIP traffic. Will this have any bad effect on sip traffic/packets? No. RTP packets are very small - much smaller than that MTU, or any reasonable MTU you could set. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 --
