[asterisk-users] SIP to IAX2 with delayed echo
A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what they wanted to use!) ___ -- 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] SIP to IAX2 with delayed echo
I'm not sure about the 3 second delay, but I've seen plenty of echo issues on Polycom phones when the gain has been changed on the handset. Check the voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're not too high. You also may want to make sure there aren't any system resource constraints such as high CPU usage or memory usage... :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - c james [EMAIL PROTECTED] wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what they wanted to use!) ___ -- 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] SIP to IAX2 with delayed echo
Just for sh1t$ and giggles, try sip to sip and drop the IAX piece. IAX2 is not all it is cracked up to be. Also, do a ping to see latency, 200ms is pretty much my standard. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Thu, Nov 20, 2008 at 12:16 PM, Tim Nelson [EMAIL PROTECTED] wrote: I'm not sure about the 3 second delay, but I've seen plenty of echo issues on Polycom phones when the gain has been changed on the handset. Check the voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're not too high. You also may want to make sure there aren't any system resource constraints such as high CPU usage or memory usage... :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - c james [EMAIL PROTECTED] wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what they wanted to use!) ___ -- 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] SIP to IAX2 with delayed echo
Tim Nelson wrote: I'm not sure about the 3 second delay, but I've seen plenty of echo issues on Polycom phones when the gain has been changed on the handset. Check the voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're not too high. You also may want to make sure there aren't any system resource constraints such as high CPU usage or memory usage... :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - c james [EMAIL PROTECTED] wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what they wanted to use!) Gains are at their default values. Definitely no problem with the resources. ___ -- 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] SIP to IAX2 with delayed echo
Steve Totaro wrote: Just for sh1t$ and giggles, try sip to sip and drop the IAX piece. IAX2 is not all it is cracked up to be. Also, do a ping to see latency, 200ms is pretty much my standard. Coming from outside the network, setting up for a couple rounds of NATting isn't going to work well. They are not seeing it between phones. Others, using the polycom phones have reported echo between two SIP on a 4ms ping trip. ___ -- 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] SIP to IAX2 with delayed echo
On Thu, Nov 20, 2008 at 1:13 PM, c james [EMAIL PROTECTED] wrote: Steve Totaro wrote: Just for sh1t$ and giggles, try sip to sip and drop the IAX piece. IAX2 is not all it is cracked up to be. Also, do a ping to see latency, 200ms is pretty much my standard. Coming from outside the network, setting up for a couple rounds of NATting isn't going to work well. They are not seeing it between phones. Others, using the polycom phones have reported echo between two SIP on a 4ms ping trip. NAT is manageable with OpenVPN and very easy. You just need a box on both sides. Also, a more difficult setup will allow SIP to work through NAT if both sides are behind a NAT. I just prefer OpenVPN because it is set it and forget it. Anyways, it is quite simple to switch to SIP to test. IAX2 has made me quite a bit of money because of it's issues, where SIP Just Works -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] SIP to IAX2 with delayed echo
There are also settings which will turn on local echo cancellation for the handset, headset and/or speaker phone. I don't recall their names at the moment. They are off by default on the handset and headset unless you're using a very recent (3.0+) SIP app. Tim Nelson wrote: I'm not sure about the 3 second delay, but I've seen plenty of echo issues on Polycom phones when the gain has been changed on the handset. Check the voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're not too high. You also may want to make sure there aren't any system resource constraints such as high CPU usage or memory usage... :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - c james [EMAIL PROTECTED] wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what they wanted to use!) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP to IAX2 with delayed echo
Simple tests. Change from the highly touted IAX2 to SIP, but before that, download X-Lite and see if you have the same delay. If you don't then look at your Polycoms, if you do, then switch to SIP. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Thu, Nov 20, 2008 at 1:39 PM, Dave Fullerton [EMAIL PROTECTED] wrote: There are also settings which will turn on local echo cancellation for the handset, headset and/or speaker phone. I don't recall their names at the moment. They are off by default on the handset and headset unless you're using a very recent (3.0+) SIP app. Tim Nelson wrote: I'm not sure about the 3 second delay, but I've seen plenty of echo issues on Polycom phones when the gain has been changed on the handset. Check the voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're not too high. You also may want to make sure there aren't any system resource constraints such as high CPU usage or memory usage... :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - c james [EMAIL PROTECTED] wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what they wanted to use!) ___ -- 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] SIP to IAX2 with delayed echo
c james wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Which end hears the echo? If it is the Polycom end, try a better quality headset with the softphone. Echo comes from analogue portions of the circuit and is usually caused at the end that doesn't hear it. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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] SIP to IAX2 with delayed echo
Ok, I'll bite, what possible IAX bugs/shortcomings/features can cause echo ? Tim. On 20 Nov 2008, at 18:47, Steve Totaro wrote: Simple tests. Change from the highly touted IAX2 to SIP, but before that, download X-Lite and see if you have the same delay. If you don't then look at your Polycoms, if you do, then switch to SIP. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Thu, Nov 20, 2008 at 1:39 PM, Dave Fullerton [EMAIL PROTECTED] wrote: There are also settings which will turn on local echo cancellation for the handset, headset and/or speaker phone. I don't recall their names at the moment. They are off by default on the handset and headset unless you're using a very recent (3.0+) SIP app. Tim Nelson wrote: I'm not sure about the 3 second delay, but I've seen plenty of echo issues on Polycom phones when the gain has been changed on the handset. Check the voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're not too high. You also may want to make sure there aren't any system resource constraints such as high CPU usage or memory usage... :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - c james [EMAIL PROTECTED] wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's what they wanted to use!) ___ -- 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] SIP to IAX2 with delayed echo
c james [EMAIL PROTECTED] writes: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Feedback from speaker to microphone. The problem is always at the end which doesn't hear it. /Benny ___ -- 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] SIP to IAX2 with delayed echo
Drew Gibson wrote: c james wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Which end hears the echo? If it is the Polycom end, try a better quality headset with the softphone. Echo comes from analogue portions of the circuit and is usually caused at the end that doesn't hear it. regards, Drew Both side are seeing it. ___ -- 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] SIP to IAX2 with delayed echo
Steve Totaro wrote: On Thu, Nov 20, 2008 at 1:13 PM, c james [EMAIL PROTECTED] wrote: Steve Totaro wrote: Just for sh1t$ and giggles, try sip to sip and drop the IAX piece. IAX2 is not all it is cracked up to be. Also, do a ping to see latency, 200ms is pretty much my standard. Coming from outside the network, setting up for a couple rounds of NATting isn't going to work well. They are not seeing it between phones. Others, using the polycom phones have reported echo between two SIP on a 4ms ping trip. NAT is manageable with OpenVPN and very easy. You just need a box on both sides. Also, a more difficult setup will allow SIP to work through NAT if both sides are behind a NAT. I just prefer OpenVPN because it is set it and forget it. Anyways, it is quite simple to switch to SIP to test. IAX2 has made me quite a bit of money because of it's issues, where SIP Just Works I'll get the network guards involved and see. ___ -- 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] SIP to IAX2 with delayed echo
Coming from outside the network, setting up for a couple rounds of NATting isn't going to work well. They are not seeing it between phones. Others, using the polycom phones have reported echo between two SIP on a 4ms ping trip. Could this be due to a purely acoustic echo within the Polycom handsets? I encountered a nasty echo / hollow sound when using a cheap USB telephone to connect to my Asterisk system (via KPhoneSI). The echoing was due to acoustic feedback - the handset body acted as a very nice channel for sound waves from the back side of the speaker down to the microphone cartridge. I opened up the handset, added some damping materials (panel- vibration-damping and soft-foam sheeting, left over from a car stereo speaker installation I did), closed it back up, and the echoing was gone. You might not notice in some calls, if the Polycom phones have silence-detection turned on for those calls and if the amount of feedback falls below the phones' silence threshold. If the phone silence-detection algorithm were turned off on other calls, the echo would then be audible. ___ -- 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] SIP to IAX2 with delayed echo
On Thu, Nov 20, 2008 at 6:27 PM, Dave Platt [EMAIL PROTECTED] wrote: Coming from outside the network, setting up for a couple rounds of NATting isn't going to work well. They are not seeing it between phones. Others, using the polycom phones have reported echo between two SIP on a 4ms ping trip. Could this be due to a purely acoustic echo within the Polycom handsets? I encountered a nasty echo / hollow sound when using a cheap USB telephone to connect to my Asterisk system (via KPhoneSI). The echoing was due to acoustic feedback - the handset body acted as a very nice channel for sound waves from the back side of the speaker down to the microphone cartridge. I opened up the handset, added some damping materials (panel- vibration-damping and soft-foam sheeting, left over from a car stereo speaker installation I did), closed it back up, and the echoing was gone. You might not notice in some calls, if the Polycom phones have silence-detection turned on for those calls and if the amount of feedback falls below the phones' silence threshold. If the phone silence-detection algorithm were turned off on other calls, the echo would then be audible. Troubleshooting is simple. Register X-Lite and see if the problem goes away. If so it is IAX2, a moving target that sometimes works OK but more often does not. If the X-Lite softphone does not show echo, then it has something to do with the Polycoms, If you still have echo, then drop IAX2, if that doesn't do it, then mess with the polycoms. When will Asterisk get VAD? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
