Re: [asterisk-users] audio glitches in conference
Jeff Brower wrote: Jonathan- How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? I just loaded the file into audacity and measured the gaps by looking at the wave form. I just went through some of the samples I recorded yesterday, and found that all the gaps are multiples of 8 samples - from 8 up to 32. I guess that's because dahdi/zaptel ticks are 8 samples each, so if it misses one, 8 samples get lost. I can't imagine that RTP is involved, since this is happening with purely local channels (just the Playback application and the eagi script) Did you measure the distance between gaps? If those distances are multiples of RTP payload length, then possibly a network latency issue is involved, but otherwise I agree with other posters, it sounds more like a timing and/or sampling synchronization problem. I just did this, and oddly enough, the distance between gaps *is* multiples of 80 - 640, 7360, 8480 samples on the recordings that I just tested. I say oddly, because there is no network latency involved - everything is local. It's playing a local audio file into a conference and recording the audio through the eagi script - nothing else. Are you handling TDM data, for example T1/E1 or ISDN? I'm not sure what TDM data is, but I'm pretty sure I'm not using it. The only telephony thing used in my application is a handful of sip connections, but these problems show up even if they aren't used, as mentioned above. Now I'm getting rather confused... why would I be seeing the RTP payload length show up up anywhere at all? Maybe asterisk uses it internally for something else? -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] audio glitches in conference
Jonathan- Jeff Brower wrote: Jonathan- How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? I just loaded the file into audacity and measured the gaps by looking at the wave form. I just went through some of the samples I recorded yesterday, and found that all the gaps are multiples of 8 samples - from 8 up to 32. I guess that's because dahdi/zaptel ticks are 8 samples each, so if it misses one, 8 samples get lost. I can't imagine that RTP is involved, since this is happening with purely local channels (just the Playback application and the eagi script) Did you measure the distance between gaps? If those distances are multiples of RTP payload length, then possibly a network latency issue is involved, but otherwise I agree with other posters, it sounds more like a timing and/or sampling synchronization problem. I just did this, and oddly enough, the distance between gaps *is* multiples of 80 - 640, 7360, 8480 samples on the recordings that I just tested. I say oddly, because there is no network latency involved - everything is local. It's playing a local audio file into a conference and recording the audio through the eagi script - nothing else. 80 samples is 10 msec of 8 kHz sampled voice. 10 msec is the default frame size in most voice systems. In some way, either via external TDM interface line clock (see comments below), or internally, Asterisk generates a 10 msec clock. Are you handling TDM data, for example T1/E1 or ISDN? I'm not sure what TDM data is, but I'm pretty sure I'm not using it. The only telephony thing used in my application is a handful of sip connections, but these problems show up even if they aren't used, as mentioned above. Ok, so you have no ISDN, T1/E1, FSX/FSO cards? These are all TDM data type interfaces. Now I'm getting rather confused... why would I be seeing the RTP payload length show up up anywhere at all? Maybe asterisk uses it internally for something else? I don't know enough about internal Asterisk timing. But I can guess that if you don't have TDM interfaces then Asterisk generates a single system clock. I don't know specifically how it does this, but anything in your config files related to clock source/generation/timing, real-time scheduling, etc is where I would look. -Jeff -- _ -- 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] audio glitches in conference
It looks to me that u are having clock synchronism problems due to the fact you are using Virtual Machine so u don't have an ISDN card generating clock. Are u using what was called ztdummie as clock source? Can't precise the name of it in chan_dahdi but u have it. What u report isn't new and is well known due to the fact u don't have a precise clock source for meetme.. You need to have chan_dahdi dummie... Hope it helps. Marco Mouta Enviada do dispositivo sem fios BlackBerry® -Original Message- From: Jeff Brower jbro...@signalogic.com Date: Wed, 24 Feb 2010 18:25:07 To: Jonathan Addlemanj...@redowl.ca Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] audio glitches in conference Jonathan- I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? -Jeff First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] audio glitches in conference
Thanks to Tzafrir for the above mentiong documentation. FYI http://docs.tzafrir.org.il/dahdi-linux/README.html A PBX system should generally have a single clock. If you are connected to a telephony provider via a digital interface (e.g: E1, T1) you should also typically use the provider's clock (as you get through the interface). Hence one important job of Asterisk is to provide timing to the PBX. DAHDI ticks once per millisecond (1000 times per second). On each tick every active DAHDI channel reads and 8 bytes of data. Asterisk also uses this for timing, through a DAHDI pseudo channel it opens. However, not all PBX systems are connected to a telephony provider via a T1 or similar connection. With an analog connection you are not synced to the other party. And some systems don't have DAHDI hardware at all. Even a digital card may be used for other uses or is simply not connected to a provider. DAHDI cards are also capable of providing timing from a clock on card. Cheap x100P clone cards are sometimes used for that purpose. If all the above fail, you can use the module dahdi_dummy to provide timing alone without needing any DAHDI hardware. It will work with most systems and kernels. You can check the DAHDI timing source with dahdi_test, which is a small utility that is included with DAHDI. It runs in cycles. In each such cycle it tries to read 8192 bytes, and sees how long it takes. If DAHDI is not loaded or you don't have the device files, it will fail immediately. If you lack a timing device it will hang forever in the first cycle. Otherwise it will just give you in each cycle the percent of how close it was. Also try running it with the option -v for a verbose output. To check the clock source that is built into dahdi_dummy, you can either look at title of its span in /proc/dahdi file for a source: in the description. Or even run: strings dahdi.ko | grep source: -- Marco Mouta On Thu, Feb 25, 2010 at 8:15 AM, marco.mo...@gmail.com wrote: It looks to me that u are having clock synchronism problems due to the fact you are using Virtual Machine so u don't have an ISDN card generating clock. Are u using what was called ztdummie as clock source? Can't precise the name of it in chan_dahdi but u have it. What u report isn't new and is well known due to the fact u don't have a precise clock source for meetme.. You need to have chan_dahdi dummie... Hope it helps. Marco Mouta Enviada do dispositivo sem fios BlackBerry® -Original Message- From: Jeff Brower jbro...@signalogic.com Date: Wed, 24 Feb 2010 18:25:07 To: Jonathan Addlemanj...@redowl.ca Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] audio glitches in conference Jonathan- I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? -Jeff First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca
Re: [asterisk-users] audio glitches in conference
marco.mo...@gmail.com wrote: It looks to me that u are having clock synchronism problems due to the fact you are using Virtual Machine so u don't have an ISDN card generating clock. Are u using what was called ztdummie as clock source? Can't precise the name of it in chan_dahdi but u have it. zt_dummy is the timing source, yes. I saw the exact same thing happening when I used app_conference though, without zt_dummy loaded at all. I also checked if I could see what source zt_dummy is using, but there's no 'source' line in /proc/zaptel/1 - just one line saying 'Span 1: ZTDUMMY/1 ZTDUMMY/1 1 ' -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] audio glitches in conference
Jeff Brower wrote: How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? I just loaded the file into audacity and measured the gaps by looking at the wave form. I just went through some of the samples I recorded yesterday, and found that all the gaps are multiples of 8 samples - from 8 up to 32. I guess that's because dahdi/zaptel ticks are 8 samples each, so if it misses one, 8 samples get lost. I can't imagine that RTP is involved, since this is happening with purely local channels (just the Playback application and the eagi script) -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] audio glitches in conference
Jonathan- How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? I just loaded the file into audacity and measured the gaps by looking at the wave form. I just went through some of the samples I recorded yesterday, and found that all the gaps are multiples of 8 samples - from 8 up to 32. I guess that's because dahdi/zaptel ticks are 8 samples each, so if it misses one, 8 samples get lost. I can't imagine that RTP is involved, since this is happening with purely local channels (just the Playback application and the eagi script) Did you measure the distance between gaps? If those distances are multiples of RTP payload length, then possibly a network latency issue is involved, but otherwise I agree with other posters, it sounds more like a timing and/or sampling synchronization problem. Are you handling TDM data, for example T1/E1 or ISDN? If so, then normally the TDM card inputs the external T1 line clock and then provides it as a master for software handling TDM data. If somehow the TDM clock is not configured this way, or the software (Asterisk) is using another clock, then effectively you have two (2) different TDM clocks and they are drifting with respect to each other. In this way you could miss bursts of samples (or get incorrect samples) as the clocks drift to be 180 out. Then you'd be Ok again for a while until it happens again, and so on... -Jeff -- _ -- 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] audio glitches in conference
I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] audio glitches in conference
Jonathan- I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording (via eagi) after the conference has been established for half an hour or so, the stream received by the eagi script delayed by about 10 seconds. How did you measure the gaps? Using signal or speech analysis software to display the recording? If you measure number of samples between the gaps, does it correspond to multiples of RTP packet payload length (for example, for 8 kHz G711 multiples of 80 samples between gaps) ? -Jeff First, the preliminaries: I'm on a debian lenny system, using the 1:1.4.21.2~dfsg-3 asterisk package. This is a dedicated server - was running xen, but I've shut down all the domU's to test if they were interfering, so now there's no sharing going on. I've been testing with a simple eagi script to grab the audio from the conference: #!/bin/sh cat /dev/fd/3 /tmp/audio.raw I've been testing it using the following dialplan extensions: [test] exten = testeagi,1,Answer exten = testeagi,n,Wait(3) exten = testeagi,n,EAGI(testeagi.sh) exten = testmeet,1,Answer exten = testmeet,n,MeetMe(testconf,1qd) exten = testsound,1,Answer exten = testsound,n,Playback(testbeep-asterisk) (testbeep is just 30 seconds of sine wave) I've been trying things like this: originate Local/testso...@test extension teste...@test The recorded audio plays back fine - no glitches. (an example is at http://www.vecotourism.org/audio17.wav) originate Local/teste...@test extension testm...@test originate Local/testso...@test extension testm...@test This does have the glitches. (an example is at http://www.vecotourism.org/audio18.wav) What could be causing this? And is there anything else I could be doing to debug it? Thanks. -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] audio glitches in conference
I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. What version of DAHDI are you running? -- _ -- 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
