Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
So, can no one give me any suggestions? Perhaps I can elaborate upon further testing and attempts to debug this tremendously frustrating problem. My softphone (typically IAXComm, but same results connecting via SIP on Xten Xlite) is installed on a P4 1.6 w/ 256 megs of RAM and an integrated sound chipset (Intel/AC97). I've had some problems with this chipset in Linux, and it doesn't support hardware mixing, so I've had to attempt to get dmix and ALSA running in an acceptable fashion; needless to say, I still have problems, and I don't know if this is related. I can record and playback my own voice and other audio (podcasts, Net radio, and music) fine with this headset (some cheap, Chinese $15 headset). However, when it comes to receiving decoded audio on the other end of a VoIP conversation, it sounds "scratchy," distorted, crackly, whatever you want to call it. It's not the clarity so much as it is the other things I just mentioned. It's very hard ot put into words, but I'm hoping *someone* can associate with my problem. To make sure it wasn't my Asterisk box that was mucking things up, I connected directly with my softphone to my outgoing VoIP terminators, voipjet and Voxee. Sure enough, it sounded *exactly* the same as it does going through my * box on the LAN and then out through the public Net to voipjet and/or Voxee. Thus, I know it is my side of the equation that is mucking things up, but I cannot for the life of me pinpoint exactly *WHERE* this is taking place. Actually, I also unplugged my headphones and plugged back in the speakers, and it sounded roughly the same, but it's harder to tell because they're not surrounding your ears, quality isn't as clear, and ambient noise can be more easily heard and is distracting. I'm thinking, at this point, that it's my sound card that's messing things up, or its configuration or something. However, can anyone explain *why* non-VoIP-conversations sound perfect on my speakers and headset, while VoIP calls sound very bad? Or, am I totally overreacting, should I expect this as the standard for softphone calls with VoIP, and should I just get a hardphone and stop worrying about it? However, the geek in me really wants to pinpoint the source of this problem! Please help, all, as this problem is occupying me for days on end and I can't get anything else done. :-) I really want to figure out why I hear scratchiness, skipping, and general lack of clarity on the other side. *Please note that I can call in to my PSTN number on the Asterisk system and hear the demo (Allison) pretty much perfectly, so it's definitely not an Asterisk problem!* Regards, Robert Geller Robert Geller wrote: UPDATE: I've been advised by users on #asterisk (IRC) that this is standard for softphones in general, and that if I were to use a hardphone, quality would be significantly better. Is this the case? Are softphones that much inferior to hardphones? That might make sense to me, as they have to go through the sound card of your computer and then out through -- and with -- all the other Net traffic from the computer you're using the softphone on. Again, is this the case, or should there be little difference between a soft- and hardphone? Also, regarding the Monitor() command, I wanted to see what the quality would be like on a recorded and played-back conversation, as I thought maybe that would clue me in on some of the problems, but it sounded pretty similar to how it sounded to me on the headset when I was talking (this was a conversation with Newegg.com's tech support). Can anyone tell me why that is? I don't really know how the Monitor() command works (I mean, I understand the concept, but not /how/ it actually goes about recording the channel(s). Would it be expected that you would hear the same quality from the other side if you listen to a Monitor()ed conversation? Thanks a lot, all. Robert Geller wrote: Hello all, I am using a headset and the X-lite softphone (sometimes I use IAXComm, but I'm having difficulties using OSS emulation with it) to connect via uLaw to my internal Asterisk server, which is a Pentium II 400 with 128 megs of RAM. After getting this headset, most or all of the echo people on the other line were complaining about is now gone, according to them. However, every five to ten seconds, I get quick "skipping" or lag on the other side, so that the person whom I'm talking to's voice sounds like it "skips a beat," analogous to when a CD you're listening to skips quickly. I don't think -- but am not positive -- that it is a question of insufficient bandwith, as I am on a Cox 5mbps/2mbps cable line that is very reliable and pretty stable. I believe I am using uLaw both to the Asterisk server /and/ from the Asterisk server to Voxee, my outgoing SIP provider/PSTN terminator. Is this a common problem? It doesn't seem like it should be, as it is a major detriment to having enjoyable, go
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
> So, can no one give me any suggestions? Perhaps I can elaborate upon > further testing and attempts to debug this tremendously frustrating problem. > > My softphone (typically IAXComm, but same results connecting via SIP on > Xten Xlite) is installed on a P4 1.6 w/ 256 megs of RAM and an > integrated sound chipset (Intel/AC97). I've had some problems with this > chipset in Linux, and it doesn't support hardware mixing, so I've had to > attempt to get dmix and ALSA running in an acceptable fashion; needless > to say, I still have problems, and I don't know if this is related. > > I can record and playback my own voice and other audio (podcasts, Net > radio, and music) fine with this headset (some cheap, Chinese $15 > headset). However, when it comes to receiving decoded audio on the other > end of a VoIP conversation, it sounds "scratchy," distorted, crackly, > whatever you want to call it. It's not the clarity so much as it is the > other things I just mentioned. It's very hard ot put into words, but I'm > hoping *someone* can associate with my problem. > > To make sure it wasn't my Asterisk box that was mucking things up, I > connected directly with my softphone to my outgoing VoIP terminators, > voipjet and Voxee. Sure enough, it sounded *exactly* the same as it does > going through my * box on the LAN and then out through the public Net to > voipjet and/or Voxee. Thus, I know it is my side of the equation that is > mucking things up, but I cannot for the life of me pinpoint exactly > *WHERE* this is taking place. > > Actually, I also unplugged my headphones and plugged back in the > speakers, and it sounded roughly the same, but it's harder to tell > because they're not surrounding your ears, quality isn't as clear, and > ambient noise can be more easily heard and is distracting. > > I'm thinking, at this point, that it's my sound card that's messing > things up, or its configuration or something. However, can anyone > explain *why* non-VoIP-conversations sound perfect on my speakers and > headset, while VoIP calls sound very bad? If I were to try to diagnose the above, I'd be using Ethereal to capture the voip packets coming from your itsp's, and analyzing that captured data to look for unusual things. If your itsp connections are sip based, ethereal has a utility to analyze/summarize some of this for you. If those connections are iax based, then you will need to analyze the packets yourself looking for unusual things. Analyzing the packets (either sip or iax) can consume a lot of time, but you really need to ensure those packets are arriving in a consistent manner, timestamps contained within the packets are consecutive and proper, packets are not arriving out of order, etc. At the same time that you're capturing those packets, use the facilities within asterisk to summarize what it thinks is going on (eg, 'iax2 show netstats'), so _that_ data can be correlated to the info derived from the packet captures. If this is a small soho * system, then run ethereal right on the same asterisk system capturing the data as it arrives at the system. Doing so will help identify any issues that you might have involving your local lan, broadband issues, etc, etc. Assuming the problem that you've described is consistent and happens on a regular basis, you don't need to collect and analyze megabytes of packet captures. Just collect a short duration sample that is assured to contain the representative packets associated with the bad audio (maybe five or ten seconds worth). If the analysis does not indicate a problem at that point, then at least you know the problem is internal to asterisk, etc. If you don't feel you have the skills or knowledge to do that analysis, then hire someone that can. Both your original post and the followup post contain a ton of adjectives and adverbs describing a technical problem, but contain little (or no) technical data (such as the results from above or results from various * show commands) that would allow anyone to comment on your problem. So, doubtful anyone is ever going to reply to such postings with anything helpful. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Rich Adamson wrote: So, can no one give me any suggestions? Perhaps I can elaborate upon further testing and attempts to debug this tremendously frustrating problem. My softphone (typically IAXComm, but same results connecting via SIP on Xten Xlite) is installed on a P4 1.6 w/ 256 megs of RAM and an integrated sound chipset (Intel/AC97). I've had some problems with this chipset in Linux, and it doesn't support hardware mixing, so I've had to attempt to get dmix and ALSA running in an acceptable fashion; needless to say, I still have problems, and I don't know if this is related. I can record and playback my own voice and other audio (podcasts, Net radio, and music) fine with this headset (some cheap, Chinese $15 headset). However, when it comes to receiving decoded audio on the other end of a VoIP conversation, it sounds "scratchy," distorted, crackly, whatever you want to call it. It's not the clarity so much as it is the other things I just mentioned. It's very hard ot put into words, but I'm hoping *someone* can associate with my problem. To make sure it wasn't my Asterisk box that was mucking things up, I connected directly with my softphone to my outgoing VoIP terminators, voipjet and Voxee. Sure enough, it sounded *exactly* the same as it does going through my * box on the LAN and then out through the public Net to voipjet and/or Voxee. Thus, I know it is my side of the equation that is mucking things up, but I cannot for the life of me pinpoint exactly *WHERE* this is taking place. Actually, I also unplugged my headphones and plugged back in the speakers, and it sounded roughly the same, but it's harder to tell because they're not surrounding your ears, quality isn't as clear, and ambient noise can be more easily heard and is distracting. I'm thinking, at this point, that it's my sound card that's messing things up, or its configuration or something. However, can anyone explain *why* non-VoIP-conversations sound perfect on my speakers and headset, while VoIP calls sound very bad? If I were to try to diagnose the above, I'd be using Ethereal to capture the voip packets coming from your itsp's, and analyzing that captured data to look for unusual things. If your itsp connections are sip based, ethereal has a utility to analyze/summarize some of this for you. If those connections are iax based, then you will need to analyze the packets yourself looking for unusual things. Analyzing the packets (either sip or iax) can consume a lot of time, but you really need to ensure those packets are arriving in a consistent manner, timestamps contained within the packets are consecutive and proper, packets are not arriving out of order, etc. At the same time that you're capturing those packets, use the facilities within asterisk to summarize what it thinks is going on (eg, 'iax2 show netstats'), so _that_ data can be correlated to the info derived from the packet captures. If this is a small soho * system, then run ethereal right on the same asterisk system capturing the data as it arrives at the system. Doing so will help identify any issues that you might have involving your local lan, broadband issues, etc, etc. Assuming the problem that you've described is consistent and happens on a regular basis, you don't need to collect and analyze megabytes of packet captures. Just collect a short duration sample that is assured to contain the representative packets associated with the bad audio (maybe five or ten seconds worth). If the analysis does not indicate a problem at that point, then at least you know the problem is internal to asterisk, etc. If you don't feel you have the skills or knowledge to do that analysis, then hire someone that can. Both your original post and the followup post contain a ton of adjectives and adverbs describing a technical problem, but contain little (or no) technical data (such as the results from above or results from various * show commands) that would allow anyone to comment on your problem. So, doubtful anyone is ever going to reply to such postings with anything helpful. Thank you very much for your response. I do acknowledge that my previous posts did not contain much technical information to speak of, but it was mainly because I wasn't/am not familiar with the Asterisk CLI and troubleshooting Asterisk problems, so I apologize for that. I did get the idea early this morning to try to analyze packets with ethereal, and I captured packets when I was made an internal IAX call to the Asterisk system (voicemail). I don't really know what to look for, but I will learn (again, I'm not very familiar with ethereal). Do you hapeople say ve any suggestions for filters to use, to evaluate possible packet loss or resending of data? Regarding the command that you suggested in the CLI, iax2 show netstats, it doesn't recognize that command or anything similar, and 'help' doesn't return anything similar that I can see (
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2", which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. I had a problem with the precise behavior you describe in your earlier mail. For me, the solution came when I discovered that my Ethernet card and my sound card were sharing an interrupt. I moved the Ethernet card around until it got a different IRQ, and the problem vanished instantly. YMMV. B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Brian Capouch wrote: Robert Geller wrote: At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2", which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. I had a problem with the precise behavior you describe in your earlier mail. For me, the solution came when I discovered that my Ethernet card and my sound card were sharing an interrupt. I moved the Ethernet card around until it got a different IRQ, and the problem vanished instantly. YMMV. B. Wow, very interesting. Thank you so much! BTW, regarding YMMV, did you have a separate, dedicated sound card? I don't -- it's integrated into my motherboard. Would this still apply? Of course, there are still ports in the back for in, out, and a mic, so it may still apply, but my Ethernet card is way down on the second-to-last PCI port, so would this still apply? Again, thanks very much for your support, and if you think this may still apply to my setup, I will definitely try what you recommended. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: Wow, very interesting. Thank you so much! BTW, regarding YMMV, did you have a separate, dedicated sound card? I don't -- it's integrated into my motherboard. Would this still apply? Of course, there are still ports in the back for in, out, and a mic, so it may still apply, but my Ethernet card is way down on the second-to-last PCI port, so would this still apply? Again, thanks very much for your support, and if you think this may still apply to my setup, I will definitely try what you recommended. Check what you get when you "cat /proc/interrupts" If you can get hold of a hardphone to play with, and it works just fine when it's on the same switch as the computer, then I would be tempted even further to suspect an interrupt-related problem as your culprit. B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Brian Capouch wrote: Robert Geller wrote: Wow, very interesting. Thank you so much! BTW, regarding YMMV, did you have a separate, dedicated sound card? I don't -- it's integrated into my motherboard. Would this still apply? Of course, there are still ports in the back for in, out, and a mic, so it may still apply, but my Ethernet card is way down on the second-to-last PCI port, so would this still apply? Again, thanks very much for your support, and if you think this may still apply to my setup, I will definitely try what you recommended. Check what you get when you "cat /proc/interrupts" If you can get hold of a hardphone to play with, and it works just fine when it's on the same switch as the computer, then I would be tempted even further to suspect an interrupt-related problem as your culprit. B. What should I be looking for in /proc/interrupts? If the first field in each row is the IRQ, I don't see any of the same numbers listed, so would that mean there are no conflicts? Earlier, I ordered a GXP-2000, which will hopefully ship from Voxilla on Tuesday, so that will be a very good indicator of just what my problem is, I think. However, in the meantime, I *really* would like to get this figured out! Leaving problems unfixed, to me, is simply a non-option. Thank you very much, again! Regards, Robert ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
> > > > > Thank you very much for your response. I do acknowledge that my previous > posts did not contain much technical information to speak of, but it was > mainly because I wasn't/am not familiar with the Asterisk CLI and > troubleshooting Asterisk problems, so I apologize for that. > > I did get the idea early this morning to try to analyze packets with > ethereal, and I captured packets when I was made an internal IAX call to > the Asterisk system (voicemail). I don't really know what to look for, > but I will learn (again, I'm not very familiar with ethereal). Do you > hapeople say ve any suggestions for filters to use, to evaluate possible > packet loss or resending of data? An important item to look at in each packet is the timestamp. In sip packets, the timestamp should be increasing by 160 for each conseq pkt. In iax packets, the timestamp should be increasing by 20 for each pkt. So if you see a timestamp of 3290 in one pkt and 3310 in the next (a diff of 20), that's good. Notice the increasing timestamp value and the diff. If pkt 3310 arrives before 3290, then something in the network is delaying the delivery of packets so as to cause them to not arrive in the proper order. If there are missing packets, then you'll see timestamps jumping by 40, 60, 80 or some other value (diff) for iax packets, or, similar for sip packets. > Regarding the command that you suggested in the CLI, iax2 show netstats, > it doesn't recognize that command or anything similar, and 'help' > doesn't return anything similar that I can see (I'm using 1.0.7 if that > helps). Since 1.0.7 is rather old (in the scheme of things), I'd suggest you install something newer to play with. There has been a ton of stuff that has changed since 1.0.7, but I don't recall if those changes would have anything to do with your problem. (I use nothing but cvs head, but I kind of keep an eye on how many changes are happening (and for what reason), and upgrade when the number of problems seem to be at a low. The 'iax2 show netstats' would have been added in a later version. > At this point, I'm thinking that it could be a matter of bad cabling or > something. The Cat5 cable that's running the 8 or so feet from my PC to > my router is homemade by me, and many people do report problems with > homemade cables. I may not have made it exactly right, or the untwisted > segment may be longer than 1/2", which supposedly causes distortion and > interference. Perhaps I ought to run out and buy a couple factory-made > cables to test the difference, if any, between them. Replacing the cable would probably be a good start since they are relatively cheap. Go buy a new one so there's no question about its quality. Also, keep the cable at least a little distance away from transformers, ballasts, and other things that tend to generate tons of electical noise. (Some desk lamps even have extremely noisy transformers or ballasts in them.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: What should I be looking for in /proc/interrupts? If the first field in each row is the IRQ, I don't see any of the same numbers listed, so would that mean there are no conflicts? Why don't you include the output in your mail? B. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
> >>> > >>> > >> Wow, very interesting. Thank you so much! BTW, regarding YMMV, did > >> you have a separate, dedicated sound card? I don't -- it's integrated > >> into my motherboard. Would this still apply? Of course, there are > >> still ports in the back for in, out, and a mic, so it may still > >> apply, but my Ethernet card is way down on the second-to-last PCI > >> port, so would this still apply? > >> > >> Again, thanks very much for your support, and if you think this may > >> still apply to my setup, I will definitely try what you recommended. > > > > > > Check what you get when you "cat /proc/interrupts" > > > > If you can get hold of a hardphone to play with, and it works just > > fine when it's on the same switch as the computer, then I would be > > tempted even further to suspect an interrupt-related problem as your > > culprit. > > > > B. > > > What should I be looking for in /proc/interrupts? If the first field in > each row is the IRQ, I don't see any of the same numbers listed, so > would that mean there are no conflicts? > > Earlier, I ordered a GXP-2000, which will hopefully ship from Voxilla on > Tuesday, so that will be a very good indicator of just what my problem > is, I think. > > However, in the meantime, I *really* would like to get this figured out! > Leaving problems unfixed, to me, is simply a non-option. You should see something like: [EMAIL PROTECTED] asterisk]# cat /proc/interrupts CPU0 0: 201066464 XT-PIC timer 1: 5421 XT-PIC i8042 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 10:1646446 XT-PIC eth0 11: 200987949 XT-PIC libata, wctdm 12: 148358 XT-PIC i8042 14: 208303 XT-PIC ide0 15:1808413 XT-PIC ide1 NMI: 0 ERR: 0 [EMAIL PROTECTED] asterisk]# In the above, the ethernet card is on interrupt 10 and my digium TDM card is on 11. Look for your ethernet in the list and see if anything is listed next to it. (In the above libata and wctdm are sharing an interrupt. You really don't want anything sharing an interupt with your ethernet card.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Brian Capouch wrote: Robert Geller wrote: What should I be looking for in /proc/interrupts? If the first field in each row is the IRQ, I don't see any of the same numbers listed, so would that mean there are no conflicts? Why don't you include the output in your mail? B. CPU0 0: 79766966IO-APIC-edge timer 1: 41185IO-APIC-edge i8042 7: 2IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 72IO-APIC-edge i8042 14: 213361IO-APIC-edge ide0 15: 876060IO-APIC-edge ide1 169: 0 IO-APIC-level uhci_hcd 177: 61956 IO-APIC-level Intel 82801BA-ICH2, eth2 185: 0 IO-APIC-level uhci_hcd 193: 2 IO-APIC-level ohci1394 201:2008201 IO-APIC-level eth0 NMI: 0 LOC: 79777609 ERR: 0 MIS: 0 Interesting. Eth0 is my wired LAN interface and eth2 is my wireless LAN interface. Usually, I disable the wireless for obvious reasons (it's installed only because I previously had a wireless solution when I couldn't wire the house, but I recently did so I now have wired), but now that it /and/ eth0 are enabled, I don't know which one applications are using! It's quite possible that the softphone is using eth2, which would probably cause problems as you previously mentioned. Well, I disabled eth2 -- since I don't use/want to use it anyway -- with ifconfig eth2 down, but /proc/interrupts still shows the same entry for eth2 sharing the interrupt, which, as you and Rich said, isn't good. Perhaps I should disable eth2, reboot, and see what /proc/interrupts is then, along with how the softphone sounds? Note that I can listen to streaming music and other files perfectly fine--does this still apply? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Rich Adamson wrote: Thank you very much for your response. I do acknowledge that my previous posts did not contain much technical information to speak of, but it was mainly because I wasn't/am not familiar with the Asterisk CLI and troubleshooting Asterisk problems, so I apologize for that. I did get the idea early this morning to try to analyze packets with ethereal, and I captured packets when I was made an internal IAX call to the Asterisk system (voicemail). I don't really know what to look for, but I will learn (again, I'm not very familiar with ethereal). Do you hapeople say ve any suggestions for filters to use, to evaluate possible packet loss or resending of data? An important item to look at in each packet is the timestamp. In sip packets, the timestamp should be increasing by 160 for each conseq pkt. In iax packets, the timestamp should be increasing by 20 for each pkt. So if you see a timestamp of 3290 in one pkt and 3310 in the next (a diff of 20), that's good. Notice the increasing timestamp value and the diff. If pkt 3310 arrives before 3290, then something in the network is delaying the delivery of packets so as to cause them to not arrive in the proper order. If there are missing packets, then you'll see timestamps jumping by 40, 60, 80 or some other value (diff) for iax packets, or, similar for sip packets. Regarding the command that you suggested in the CLI, iax2 show netstats, it doesn't recognize that command or anything similar, and 'help' doesn't return anything similar that I can see (I'm using 1.0.7 if that helps). Since 1.0.7 is rather old (in the scheme of things), I'd suggest you install something newer to play with. There has been a ton of stuff that has changed since 1.0.7, but I don't recall if those changes would have anything to do with your problem. (I use nothing but cvs head, but I kind of keep an eye on how many changes are happening (and for what reason), and upgrade when the number of problems seem to be at a low. The 'iax2 show netstats' would have been added in a later version. At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2", which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. Replacing the cable would probably be a good start since they are relatively cheap. Go buy a new one so there's no question about its quality. Also, keep the cable at least a little distance away from transformers, ballasts, and other things that tend to generate tons of electical noise. (Some desk lamps even have extremely noisy transformers or ballasts in them.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Your advice was *extremely* helpful. It seems that I learn something new each time I read you all's posts. To me, it looked like each packet was correctly sent at the right interval each time, but I didn't evaluate each one. However, the general trend is that there seems to be no packet loss or resending. I could buy another cable as well, just to be safe, but it seems to me the potential IRQ conflict is more the more likely problem--of course, even when I "ifconfig eth2 down"ed the interface, it still showed up in /proc/interrupts -- does bringing the interface down not completely bring it down? Should I permanently disable it and reboot? Again, thank you very much for your ongoing help; I feel like I'm paying (or ought to) for professional support here. :-) Regards, Robert Geller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: Rich Adamson wrote: Thank you very much for your response. I do acknowledge that my previous posts did not contain much technical information to speak of, but it was mainly because I wasn't/am not familiar with the Asterisk CLI and troubleshooting Asterisk problems, so I apologize for that. I did get the idea early this morning to try to analyze packets with ethereal, and I captured packets when I was made an internal IAX call to the Asterisk system (voicemail). I don't really know what to look for, but I will learn (again, I'm not very familiar with ethereal). Do you hapeople say ve any suggestions for filters to use, to evaluate possible packet loss or resending of data? An important item to look at in each packet is the timestamp. In sip packets, the timestamp should be increasing by 160 for each conseq pkt. In iax packets, the timestamp should be increasing by 20 for each pkt. So if you see a timestamp of 3290 in one pkt and 3310 in the next (a diff of 20), that's good. Notice the increasing timestamp value and the diff. If pkt 3310 arrives before 3290, then something in the network is delaying the delivery of packets so as to cause them to not arrive in the proper order. If there are missing packets, then you'll see timestamps jumping by 40, 60, 80 or some other value (diff) for iax packets, or, similar for sip packets. Regarding the command that you suggested in the CLI, iax2 show netstats, it doesn't recognize that command or anything similar, and 'help' doesn't return anything similar that I can see (I'm using 1.0.7 if that helps). Since 1.0.7 is rather old (in the scheme of things), I'd suggest you install something newer to play with. There has been a ton of stuff that has changed since 1.0.7, but I don't recall if those changes would have anything to do with your problem. (I use nothing but cvs head, but I kind of keep an eye on how many changes are happening (and for what reason), and upgrade when the number of problems seem to be at a low. The 'iax2 show netstats' would have been added in a later version. At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2", which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. Replacing the cable would probably be a good start since they are relatively cheap. Go buy a new one so there's no question about its quality. Also, keep the cable at least a little distance away from transformers, ballasts, and other things that tend to generate tons of electical noise. (Some desk lamps even have extremely noisy transformers or ballasts in them.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Your advice was *extremely* helpful. It seems that I learn something new each time I read you all's posts. To me, it looked like each packet was correctly sent at the right interval each time, but I didn't evaluate each one. However, the general trend is that there seems to be no packet loss or resending. I could buy another cable as well, just to be safe, but it seems to me the potential IRQ conflict is more the more likely problem--of course, even when I "ifconfig eth2 down"ed the interface, it still showed up in /proc/interrupts -- does bringing the interface down not completely bring it down? Should I permanently disable it and reboot? Again, thank you very much for your ongoing help; I feel like I'm paying (or ought to) for professional support here. :-) Regards, Robert Geller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I modprobe -r'ed prism54, the wireless kernel modules for my card, and here's what /proc/interrupts looks like: [EMAIL PROTECTED]:~/torrents$ cat /proc/interrupts CPU0 0: 81981003IO-APIC-edge timer 1: 49755IO-APIC-edge i8042 7: 2IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 499429IO-APIC-edge i8042 14: 224679IO-APIC-edge ide0 15: 900392IO-APIC-edge ide1 169: 0 IO-APIC-level uhci_hcd 177: 67917 IO-APIC
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Robert Geller wrote: Robert Geller wrote: Rich Adamson wrote: Thank you very much for your response. I do acknowledge that my previous posts did not contain much technical information to speak of, but it was mainly because I wasn't/am not familiar with the Asterisk CLI and troubleshooting Asterisk problems, so I apologize for that. I did get the idea early this morning to try to analyze packets with ethereal, and I captured packets when I was made an internal IAX call to the Asterisk system (voicemail). I don't really know what to look for, but I will learn (again, I'm not very familiar with ethereal). Do you hapeople say ve any suggestions for filters to use, to evaluate possible packet loss or resending of data? An important item to look at in each packet is the timestamp. In sip packets, the timestamp should be increasing by 160 for each conseq pkt. In iax packets, the timestamp should be increasing by 20 for each pkt. So if you see a timestamp of 3290 in one pkt and 3310 in the next (a diff of 20), that's good. Notice the increasing timestamp value and the diff. If pkt 3310 arrives before 3290, then something in the network is delaying the delivery of packets so as to cause them to not arrive in the proper order. If there are missing packets, then you'll see timestamps jumping by 40, 60, 80 or some other value (diff) for iax packets, or, similar for sip packets. Regarding the command that you suggested in the CLI, iax2 show netstats, it doesn't recognize that command or anything similar, and 'help' doesn't return anything similar that I can see (I'm using 1.0.7 if that helps). Since 1.0.7 is rather old (in the scheme of things), I'd suggest you install something newer to play with. There has been a ton of stuff that has changed since 1.0.7, but I don't recall if those changes would have anything to do with your problem. (I use nothing but cvs head, but I kind of keep an eye on how many changes are happening (and for what reason), and upgrade when the number of problems seem to be at a low. The 'iax2 show netstats' would have been added in a later version. At this point, I'm thinking that it could be a matter of bad cabling or something. The Cat5 cable that's running the 8 or so feet from my PC to my router is homemade by me, and many people do report problems with homemade cables. I may not have made it exactly right, or the untwisted segment may be longer than 1/2", which supposedly causes distortion and interference. Perhaps I ought to run out and buy a couple factory-made cables to test the difference, if any, between them. Replacing the cable would probably be a good start since they are relatively cheap. Go buy a new one so there's no question about its quality. Also, keep the cable at least a little distance away from transformers, ballasts, and other things that tend to generate tons of electical noise. (Some desk lamps even have extremely noisy transformers or ballasts in them.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Your advice was *extremely* helpful. It seems that I learn something new each time I read you all's posts. To me, it looked like each packet was correctly sent at the right interval each time, but I didn't evaluate each one. However, the general trend is that there seems to be no packet loss or resending. I could buy another cable as well, just to be safe, but it seems to me the potential IRQ conflict is more the more likely problem--of course, even when I "ifconfig eth2 down"ed the interface, it still showed up in /proc/interrupts -- does bringing the interface down not completely bring it down? Should I permanently disable it and reboot? Again, thank you very much for your ongoing help; I feel like I'm paying (or ought to) for professional support here. :-) Regards, Robert Geller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I modprobe -r'ed prism54, the wireless kernel modules for my card, and here's what /proc/interrupts looks like: [EMAIL PROTECTED]:~/torrents$ cat /proc/interrupts CPU0 0: 81981003IO-APIC-edge timer 1: 49755IO-APIC-edge i8042 7: 2IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 499429IO-APIC-edge i8042 14: 224679IO-APIC-edge ide0 15: 900392IO-APIC-edge ide1 169: 0 IO-APIC-level uhci_hc
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
> Sorry to write so many consecutive messages in such a short period of > time, but this problem is really bugging me as it has been going on for > days. > > When I look in Ethereal, there are actually "two calls" going on -- in > this particular call, Source call #4 and Source call #10318, #4 coming > from the asterisk server and the other one coming from my computer to > the Asterisk server. I don't know why there are two separate "calls," > but perhaps one of you do. Its not really two separate calls; its the transmit leg and the receive leg of the same call. > Anyhow, source call #10318 seems fine, > sending a new packet every 20 ms pretty much perfectly and all (although > I do see now that one packet has a timestamp of 33080 and the next has > one of 35060 -- is this something to be concerned about? The diff is 1980, which essentially suggests there were 99 iax packets missing between those two timestamps. (Should note that timestamps miscalculations have been an issue in the iax2 source code, but I don't remember if some of the fixes were before or after the version of code your running. That really was the basis for suggesting a code upgrade.) If 99 iax2 packets are actually missing, you _would_ have a problem with the audio quality. > it doesn't seem > widespread). However, call #4 seems to send every 20 ms, but then there > will be a pause or something in sending, in between which there will be > more packets from source call #10318 which are sent pretty much OK. Keep in mind that you're looking at a full duplex flow of packets in and out. The fact that packets are not exactly one transmit for one every one received is not as important as identifying missing packets in the form of large jumps in timestamp values. > Then, the next packet for source call #4 will have a timestamp of > something like 33540, exactly 200 ms after the previous packet from > source call #10318. However, the next packet for SC (source call) #10318 > increments 20 ms like it should. Every single packet then on (in this > capture, I recorded about 1500 packets) sends perfectly. iax2.rrdropped, > iax2.rrjitter, and iax2.iax.rrloss returned only 2 packets--the same > two, in the middle of the 1500 packets. So, out of 1500, these are the > only two that seem to have problems. Asterisk can generally handle a couple of missing 20 millisecond packets, but it can't compensate (or cover up) hugh timestamp jumps. Might consider doing another ethereal run or two and see if you can reach a conclusion that the timestamp jumps are in fact associated with the choppy audio. If you can reach that conclusion, the next step is to isolate why; is it bad code or the network that's causing the issue? I don't recall from your previous postings, but could you repeat what exact code versions are running on "my computer" and the remote "asterisk server"? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
Rich Adamson wrote: Sorry to write so many consecutive messages in such a short period of time, but this problem is really bugging me as it has been going on for days. When I look in Ethereal, there are actually "two calls" going on -- in this particular call, Source call #4 and Source call #10318, #4 coming from the asterisk server and the other one coming from my computer to the Asterisk server. I don't know why there are two separate "calls," but perhaps one of you do. Its not really two separate calls; its the transmit leg and the receive leg of the same call. Oh, ok, that makes sense. Anyhow, source call #10318 seems fine, sending a new packet every 20 ms pretty much perfectly and all (although I do see now that one packet has a timestamp of 33080 and the next has one of 35060 -- is this something to be concerned about? The diff is 1980, which essentially suggests there were 99 iax packets missing between those two timestamps. (Should note that timestamps miscalculations have been an issue in the iax2 source code, but I don't remember if some of the fixes were before or after the version of code your running. That really was the basis for suggesting a code upgrade.) If 99 iax2 packets are actually missing, you _would_ have a problem with the audio quality. Perhaps I read that wrong, as the different filters don't seem to show those packets as lost or jittered. Plus, this crackle happens fairly often, so I don't know if it's any indication of dropped packets (or the root of the problem, at least), but then again, I'm not sure. I will try taking another sample. it doesn't seem widespread). However, call #4 seems to send every 20 ms, but then there will be a pause or something in sending, in between which there will be more packets from source call #10318 which are sent pretty much OK. Keep in mind that you're looking at a full duplex flow of packets in and out. The fact that packets are not exactly one transmit for one every one received is not as important as identifying missing packets in the form of large jumps in timestamp values. Yeah, I see what you're saying. Then, the next packet for source call #4 will have a timestamp of something like 33540, exactly 200 ms after the previous packet from source call #10318. However, the next packet for SC (source call) #10318 increments 20 ms like it should. Every single packet then on (in this capture, I recorded about 1500 packets) sends perfectly. iax2.rrdropped, iax2.rrjitter, and iax2.iax.rrloss returned only 2 packets--the same two, in the middle of the 1500 packets. So, out of 1500, these are the only two that seem to have problems. Asterisk can generally handle a couple of missing 20 millisecond packets, but it can't compensate (or cover up) hugh timestamp jumps. Might consider doing another ethereal run or two and see if you can reach a conclusion that the timestamp jumps are in fact associated with the choppy audio. If you can reach that conclusion, the next step is to isolate why; is it bad code or the network that's causing the issue? I don't recall from your previous postings, but could you repeat what exact code versions are running on "my computer" and the remote "asterisk server"? By code versions, do you mean what OS I'm running on my computer? I'm running Debian etch (testing) and ethereal 0.10.12. On my Asterisk system, I'm also running Debian etch, with Asterisk 1.0.7 (debian's "testing" version of asterisk). Would you then suggest that I should upgrade to a later version? Which one? 1.0.9? Or are even the BETAs (1.2.x) usable? I'm not sure if it's just psychological at this point, but the crackle seems to have faded a bit; it doesn't sound as harsh or prevalent as it did before. I moved my power strip further away from my computer, and I'm not sure if this made a difference or not, but it doesn't sound *as bad*. Note that it's still bad enough that I would take a land line over it any day, and it's certainly and by all means abnormal (thus, absolutely worth resolving). I did listen to some sample sounds on Cisco's website -- http://www.cisco.com/en/US/tech/tk652/tk698/technologies_white_paper09186a00801545e4.shtml#crackle -- and determined that my symptom is most like their posted "crackling" audio quality symptom, if this helps any more. However, /their/ crackling example is much harsher, louder, and more annoying than mine. Nonetheless, I believe mine more closely resembles that sample than any of the others -- it's just less severe. It could be a big transformer that's near the Ethernet card, but I believe my strip is now a pretty standard distance away from my Ethernet card; I measured it, and it's roughly 16 inches from the Ethernet card itself. It doesn't seem like it will that much farther, but is that too close? Plus, if it is electrical interference, ethernet being digital and all, wouldn't the potential interferen
Re: [Asterisk-Users] Distortion/crackling/skipping problems on outgoing calls -- please help!!!
> Perhaps I read that wrong, as the different filters don't seem to show > those packets as lost or jittered. Plus, this crackle happens fairly > often, so I don't know if it's any indication of dropped packets (or the > root of the problem, at least), but then again, I'm not sure. I will try > taking another sample. Send me a copy of an ethereal trace off list and I'll take a look at it. When you do that, let me know what IP address is what in that trace so I have some clue what I'm looking at. > Would you then suggest that I should upgrade to a later version? Which > one? 1.0.9? Or are even the BETAs (1.2.x) usable? I'm not sure what the latest stable version is since I don't pay much attention to it. I stay with the cvs head keeping a backup copy of the previous working code on my system in case I have issues with whatever I check out. If v1.0.9 is the latest stable, then use it. You have kind of skipped over exactly what "my computer" happens to be. Specifically, tell us what O/S, what software are you running on that system that is communicating with iax2, etc. Also, when you refer to your Asterisk system, is that system on your local network or located somewhere else? > It could be a big transformer that's near the Ethernet card, but I > believe my strip is now a pretty standard distance away from my Ethernet > card; I measured it, and it's roughly 16 inches from the Ethernet card > itself. It doesn't seem like it will that much farther, but is that too > close? If external electrical noise from a transformer is impacting your ethernet cable, it would impact your music and other things as much as it would iax2. So that's probably not an issue. Power strips by themselves do not generate electrical noise, so that's a non-issue anyway. > BTW, just to reassure you all that my ethernet is fine, here are the > results of an extremely fast (ping -i 0.0005) ping to the asterisk server: > > --- 192.168.2.7 ping statistics --- > 10758 packets transmitted, 10758 received, 0% packet loss, time 28976ms > rtt min/avg/max/mdev = 0.106/0.118/10.779/0.105 ms, pipe 2, ipg/ewma > 2.693/0.118 ms > > i don't think my ethernet is flawed at all. i did several of those > tests, by the way. Okay, send me a reasonable ethereal trace and I'll take a look at it. Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
