[asterisk-users] Call quality - how to debug
Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, Thanks Adrian ___ -- 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] Call quality - how to debug
On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I’ve a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I’ve one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I’ve turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I’m at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I’m not sure what that will achieve. I’m going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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] Call quality - how to debug
- Steve Howes st...@geekinter.net wrote: On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I’ve a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I’ve one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I’ve turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I’m at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I’m not sure what that will achieve. I’m going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ Physical or virtualised server ? Best Regards, -- SplatNIX IT Services :: Innovation through collaboration ___ -- 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] Call quality - how to debug
Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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] Call quality - how to debug
I don't need QoS. The voice network here is seperated from the PC LAN physically (2 separate switches), by design. So theres no web browsing etc on that 2mb circuit. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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 ___ -- 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] Call quality - how to debug
2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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 ___ -- 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] Call quality - how to debug
Scratch that, my inventory tool says the system has 256Mb not 1Gb. I wonder if a memory upgrade would help it out... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 02 June 2009 14:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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 ___ -- 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] Call quality - how to debug
Do you have any idea the number of bugs that have been fixed since 1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this. On 06/02/2009 08:58 AM, Adrian Marsh wrote: Hi, It's a 2mb dedicated leased fibre line, with50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, ___ -- 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] Call quality - how to debug
Yeah, I know, but when I last tried an upgrade to 1.4.18 it broke the whole IAX connectivity and I was forced to drop back. I'll go: 1) Memory upgrade first 2) Clone the machine, and upgrade to latest 1.4.x However - my question would still stand, how exactly would I be able to debug whats going on in the RTP stream? And why its stuttering (sometimes halfway through a call). Any tips or tricks for actually debugging within Asterisk ? Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick Hartman Sent: 02 June 2009 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Do you have any idea the number of bugs that have been fixed since 1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this. On 06/02/2009 08:58 AM, Adrian Marsh wrote: Hi, It's a 2mb dedicated leased fibre line, with50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, ___ -- 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] Call quality - how to debug
On Tue, 2009-06-02 at 15:49 +0100, Adrian Marsh wrote: However - my question would still stand, how exactly would I be able to debug whats going on in the RTP stream? And why its stuttering (sometimes halfway through a call). Any tips or tricks for actually debugging within Asterisk ? Wireshark has a lot of RTP tools for looking at the latency and jitter and dropped packets on the line, which are the most common problems I find when helping people diagnose poor audio connections. It won't tell you what is *causing* the problem, but it will help you know what the problem actually is. From there, you can start to track down the source of the problem one network segment at a time. For example... is the poor audio being caused by network problems between the phone and Asterisk, or between Asterisk and your upstream provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Call quality - how to debug
On 2 Jun 2009, at 14:14, Adrian Marsh wrote: I’m at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I’m not sure what that will achieve. Why not just tcpdump on the asterisk box then load it into wireshark? ___ -- 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] Call quality - how to debug
I think you're overlooking your internet uplink, which is what I'm talking about: snip Inbound calls are via in IAX trunk from Gradwell. /snip You certainly DO need QOS to maintain call quality over the INTERNET link. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug I don't need QoS. The voice network here is seperated from the PC LAN physically (2 separate switches), by design. So theres no web browsing etc on that 2mb circuit. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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 ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] Call quality - how to debug
Unless I've misunderstood and you're not running ANYTHING but voice over that internet uplink? snip So theres no web browsing etc on that 2mb circuit. /snip In which case, I stand corrected and you don't need QOS. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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 ___ -- 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 ___ -- 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] Call quality - how to debug
Hi Dave, You're quite right, it's a dedicated down and uplink to my ISP, and Gradwell also has fibre connection into that ISP (so short hop to them) The reason I don't think it's the fiber link, is that Asterisk recorded the conversation as two channels. IN (from Gradwell), and OUT (from the Cisco phone, that's on the same LAN as the asterisk server). And I hear distortion on both sides, at the same time. As thats what asterisk hears, and that part of the call is a same-LAN RTP stream, pre-ISP, then that's why I don't think it's the IAX link. That said, I've not got complaints from users making internal calls. So my thinking was maybe its an IAX/SIP conversion thing As a test, I've switched my account, and the problem account to inbound SIP, to see if that makes a difference. That makes it 100% SIP. Next step, memory upgrade and the A*k upgrade. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 16:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug Unless I've misunderstood and you're not running ANYTHING but voice over that internet uplink? snip So theres no web browsing etc on that 2mb circuit. /snip In which case, I stand corrected and you don't need QOS. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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
Re: [asterisk-users] Call quality - how to debug
Hi Steve, Mainly because, if it were a CPU utilisation issue, then putting an extra load on the server because of tcpdump isn't going to help. If I go that route then I'll port mirror on the switch. But thanks for the reply, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 16:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. Why not just tcpdump on the asterisk box then load it into wireshark? ___ -- 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
