[Asterisk-Users] Echo Cancellation Feature
I do feel the echo cancellation does need some work. Currently, other than listening to users, there is no way to benchmark or trouble shoot echo problems. We find that 2 to 3 out of every 20 calls will experience echo. While echo is a problem that naturally occurs from SIP - PSTN and vice versa, I am still baffled by the fact that the cancellation works randomly. When doing a zap show channel X, it will also report that the cancellation is still on. We experience the most echo with a T100P -- Adtran TA 750 FXO modules. While I understand these do not have impedence matching, I wonder to myself why echo cancellation works sometimes, and others not. Looking at Network util, processor util, and memory utilization, they do not provide any clear indication as to why /when it occurs. Is there anything more that can be done to debug echo cancellation, and further are other users experiencing this random echo. I know it was discussed before, but the support folks at digium aren't able to offer anymore help. Asterisk is truly a great piece of open soruce software, and I commend the authors/contributors for their hardware and attention to detail. We're just a few items a way from making this thing absolutely kill the traditional PBX market :). Long live Asterisk, Brent On Thu, 22 Apr 2004, David J Carter wrote: I have my RX at 4.0 ant TX at 8.0, I get slight echo for the first 5-6 seconds then all OK. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Stenton Sent: 22 April 2004 17:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] inbound calls better quality than outbound calls on X100P I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is wrong, I'm using the latest zaptel and asterisk from the cvs head as of today. Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Echo Cancellation Feature
I would also offer feedback that we too have random calls with echo on our end, that can't be traced to a reproducible event. It's very odd and can be frustrating, as it's a big distraction for those that don't know better. Like a bad cell phone connection when you hear yourself talk. For us, this happens in a pure SIP environment on a network switch dedicated to Asterisk, a T1 PRI and 18 SIP handsets. HTH, Ryan On Apr 22, 2004, at 1:37 PM, Brent Franks wrote: I do feel the echo cancellation does need some work. Currently, other than listening to users, there is no way to benchmark or trouble shoot echo problems. We find that 2 to 3 out of every 20 calls will experience echo. While echo is a problem that naturally occurs from SIP - PSTN and vice versa, I am still baffled by the fact that the cancellation works randomly. snip ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Echo Cancellation Feature
I do feel the echo cancellation does need some work. Currently, other than listening to users, there is no way to benchmark or trouble shoot echo problems. Sure there are, it's just that 99% of the asterisk implementors don't have the test equipment to do it, and a good share probably wouldn't know how to do it if they had access to the equipment. We find that 2 to 3 out of every 20 calls will experience echo. While echo is a problem that naturally occurs from SIP - PSTN and vice versa, I am still baffled by the fact that the cancellation works randomly. When doing a zap show channel X, it will also report that the cancellation is still on. We experience the most echo with a T100P -- Adtran TA 750 FXO modules. While I understand these do not have impedence matching, I wonder to myself why echo cancellation works sometimes, and others not. Looking at Network util, processor util, and memory utilization, they do not provide any clear indication as to why /when it occurs. Not likely to have any impact whatsoever. Is there anything more that can be done to debug echo cancellation, and further are other users experiencing this random echo. I know it was discussed before, but the support folks at digium aren't able to offer anymore help. You've probably read enough from previous postings to know there are several different locations within an end-to-end voice call where echo can creap into a system. In very general terms, any place where an end-to-end channel incures a two-wire to four-wire conversion (whether done in hardware or software), echo can creap in due to lots of different reasons. Since asterisk provides us with lots of configuration choices, hardly any two systems are alike. Therefore, don't know that anyone is going to write * code anytime soon to correct something that can't be pointed to, etc. Someone mentioned they have echo on sip to sip calls (presumably the call was processed by a single * system). If they do, the problem is likely in the sip phone as there are no echo cancallation needs in that four-wire end-to-end call from an * perspective. I've got a fair amount of test equipment (and 20+ years telephony background), and am planning to assemble a document identifying some of the pstn issues, level settings, and other things impacting a reasonable system implementation. Unless someone wants to UPS a transmission test set to me quickly, the document won't be completed for several weeks. (The only test set I have access to will not be released for a couple of weeks due to classes, etc.) I'm also expecting these tests to point out a number of other transmission issues within asterisk that we'll get documented with real numbers, etc. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Echo Cancellation Feature
snip I've got a fair amount of test equipment (and 20+ years telephony background), and am planning to assemble a document identifying some of the pstn issues, level settings, and other things impacting a reasonable system implementation. Unless someone wants to UPS a transmission test set to me quickly, the document won't be completed for several weeks. (The only test set I have access to will not be released for a couple of weeks due to classes, etc.) I'm also expecting these tests to point out a number of other transmission issues within asterisk that we'll get documented with real numbers, etc. Rich ___ Rich, One suggestion I would like to make, is where possible, tell us how to replicate the tests as best we can if we don't have the proper equipment. I'd venture to say most of us have a good, fairly sensitive, digital VOM. I know not all tests can be made with that, but I'm sure some of them can. There are fairly accurate tone generator programs that work with a sound card, same with data decoding. My point is, as you said before, not everyone has a multi-thousand dollar test set, but would still like to do what they can to properly implement a telephone system. Thanks, Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Echo Cancellation Feature
At 02:30 PM 4/22/2004, you wrote: I would also offer feedback that we too have random calls with echo on our end, that can't be traced to a reproducible event. It's very odd and can be frustrating, as it's a big distraction for those that don't know better. Like a bad cell phone connection when you hear yourself talk. For us, this happens in a pure SIP environment on a network switch dedicated to Asterisk, a T1 PRI and 18 SIP handsets. HTH, Ryan We have three Cisco 7940 SIP phones and 1 POTS phone connected to our * server with TDM10B fxs card. Our * server is connected to the pstn with 3 X100P cards. We have similar echo problems but only on our SIP phones. We do not have any echo problems with the POTS phone. We just purchased a Polycom IP 500 SIP phone to test but I expect similar echo problems. The recent thread which addressed milliwatt generators and gain adjustments helped to reduce our echo but the thread was never completed. We are not sure how to test the TX gain adjustment and where on the graph to set the RX gain when using a milliwatt generator tone. Tom Tom On Apr 22, 2004, at 1:37 PM, Brent Franks wrote: I do feel the echo cancellation does need some work. Currently, other than listening to users, there is no way to benchmark or trouble shoot echo problems. We find that 2 to 3 out of every 20 calls will experience echo. While echo is a problem that naturally occurs from SIP - PSTN and vice versa, I am still baffled by the fact that the cancellation works randomly. snip ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Echo Cancellation Feature
We have three Cisco 7940 SIP phones and 1 POTS phone connected to our * server with TDM10B fxs card. Our * server is connected to the pstn with 3 X100P cards. We have similar echo problems but only on our SIP phones. We do not have any echo problems with the POTS phone. We just purchased a Polycom IP 500 SIP phone to test but I expect similar echo problems. Don't expect the IP 500's to do anything as you stated, we are using these now and are having the problem I described earlier with these phones. - B ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Echo Cancellation Feature
The single most usefull tool that anyone outside telco consultants is likely to have is ztmonitor. If you do a ztmonitor [channel number] -v you will get a visual of the sound strengths and it's pretty easy to see when rx or tx are out of balance... Now, if only that would help fix the low-level static noise I have on the x100p, that would be great. Chris. On Thu, 22 Apr 2004, Rich Adamson wrote: I do feel the echo cancellation does need some work. Currently, other than listening to users, there is no way to benchmark or trouble shoot echo problems. Sure there are, it's just that 99% of the asterisk implementors don't have the test equipment to do it, and a good share probably wouldn't know how to do it if they had access to the equipment. We find that 2 to 3 out of every 20 calls will experience echo. While echo is a problem that naturally occurs from SIP - PSTN and vice versa, I am still baffled by the fact that the cancellation works randomly. When doing a zap show channel X, it will also report that the cancellation is still on. We experience the most echo with a T100P -- Adtran TA 750 FXO modules. While I understand these do not have impedence matching, I wonder to myself why echo cancellation works sometimes, and others not. Looking at Network util, processor util, and memory utilization, they do not provide any clear indication as to why /when it occurs. Not likely to have any impact whatsoever. Is there anything more that can be done to debug echo cancellation, and further are other users experiencing this random echo. I know it was discussed before, but the support folks at digium aren't able to offer anymore help. You've probably read enough from previous postings to know there are several different locations within an end-to-end voice call where echo can creap into a system. In very general terms, any place where an end-to-end channel incures a two-wire to four-wire conversion (whether done in hardware or software), echo can creap in due to lots of different reasons. Since asterisk provides us with lots of configuration choices, hardly any two systems are alike. Therefore, don't know that anyone is going to write * code anytime soon to correct something that can't be pointed to, etc. Someone mentioned they have echo on sip to sip calls (presumably the call was processed by a single * system). If they do, the problem is likely in the sip phone as there are no echo cancallation needs in that four-wire end-to-end call from an * perspective. I've got a fair amount of test equipment (and 20+ years telephony background), and am planning to assemble a document identifying some of the pstn issues, level settings, and other things impacting a reasonable system implementation. Unless someone wants to UPS a transmission test set to me quickly, the document won't be completed for several weeks. (The only test set I have access to will not be released for a couple of weeks due to classes, etc.) I'm also expecting these tests to point out a number of other transmission issues within asterisk that we'll get documented with real numbers, etc. Rich -- chris maresca senior partner - www.olliancegroup.com linux, up 17 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Echo Cancellation Feature
I've got a really cheap analog phone connected to a Sipura SIP adaptor, and have zero echo problems... Just static problems, but it may be related. Chris. On Thu, 22 Apr 2004, Brent Franks wrote: We have three Cisco 7940 SIP phones and 1 POTS phone connected to our * server with TDM10B fxs card. Our * server is connected to the pstn with 3 X100P cards. We have similar echo problems but only on our SIP phones. We do not have any echo problems with the POTS phone. We just purchased a Polycom IP 500 SIP phone to test but I expect similar echo problems. Don't expect the IP 500's to do anything as you stated, we are using these now and are having the problem I described earlier with these phones. - B -- chris maresca senior partner - www.olliancegroup.com linux, up 17 days ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Echo Cancellation Feature
snip I've got a fair amount of test equipment (and 20+ years telephony background), and am planning to assemble a document identifying some of the pstn issues, level settings, and other things impacting a reasonable system implementation. Unless someone wants to UPS a transmission test set to me quickly, the document won't be completed for several weeks. (The only test set I have access to will not be released for a couple of weeks due to classes, etc.) I'm also expecting these tests to point out a number of other transmission issues within asterisk that we'll get documented with real numbers, etc. snip One suggestion I would like to make, is where possible, tell us how to replicate the tests as best we can if we don't have the proper equipment. I'd venture to say most of us have a good, fairly sensitive, digital VOM. I know not all tests can be made with that, but I'm sure some of them can. There are fairly accurate tone generator programs that work with a sound card, same with data decoding. My point is, as you said before, not everyone has a multi-thousand dollar test set, but would still like to do what they can to properly implement a telephone system. I'm about 95% confident I can measure and describe several different issues that truly have been impacting interfaces to pstn lines, etc. But, I need to validate the steps before running off at the mouth with misrepresentations, etc. I don't believe these issues are asterisk related at all, but rather outside influences that many are feeling but can't see (or deal with). Assuming I'm correct, the instrument needed to identify at least some of these issues retails for about $300 US. Example, www.triplett.com click on Test Equipment, Telco Testers, Model 2 to 7, don't know for sure as yet. Lots of other venders as well. To prove the point (and write the documentation), several other pieces of test equipment will be likely for me, but not needed by you. The unit I'm borrowing sells for over $3,000 but does lots of other stuff not needed in typical asterisk deployments. Just about every telephone installer in the US (there are exceptions) is carrying something similar to the above. They use it to measure levels to the milliwatt generator, and they use the noise measurement side to the quiet termination. The milliwatt generator and quiet termination are often times implemented in the same piece of telco hardware, but they have different telephone numbers assigned to them. (Don't think we have a quiet termination in asterisk as yet, but should be easy to implement if it is actually needed.) The standard VOM isn't going to cut it for these tests as they are no where near sensitive enough to accurately measure levels, noise, AC induction, etc. But, if we're all going to play in the telephony business then we better buy (and understand how to use) the tools necessary to play in that business. I think I can help. I'm kind of thinking that if this can be described accurately, I'd guess one of the developers can add some code to help measure/identify the issues. If that impression is correct, then the test set won't be required at all. Let's see how it goes! Hopefully I can help that process and understanding even though I'm not a coder. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Echo Cancellation Feature
At 02:30 PM 4/22/2004, you wrote: I would also offer feedback that we too have random calls with echo on our end, that can't be traced to a reproducible event. It's very odd and can be frustrating, as it's a big distraction for those that don't know better. Like a bad cell phone connection when you hear yourself talk. For us, this happens in a pure SIP environment on a network switch dedicated to Asterisk, a T1 PRI and 18 SIP handsets. HTH, Ryan We have three Cisco 7940 SIP phones and 1 POTS phone connected to our * server with TDM10B fxs card. Our * server is connected to the pstn with 3 X100P cards. We have similar echo problems but only on our SIP phones. We do not have any echo problems with the POTS phone. We just purchased a Polycom IP 500 SIP phone to test but I expect similar echo problems. The recent thread which addressed milliwatt generators and gain adjustments helped to reduce our echo but the thread was never completed. We are not sure how to test the TX gain adjustment and where on the graph to set the RX gain when using a milliwatt generator tone. The thread truly has not been completed and will likely take a few more weeks to do it. I do believe we'll get to the bottom of it though, just hand tight. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
