Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: ReetP on February 09, 2010, 05:46:08 PM
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Have a SME box currently running Asterisk 1.4.25.1-78 and sail-2.4.1-2 and all works very nicely integrating well via VOIP with my Telrad exchange
The only problem I have (well, I don't but people I speak to do) is that the other end can hear an echo of their own voice. I can hear them clearly and get no echo my end.
The Asterisk box sits next to the Telrad box physically and on the same network. sip show peers shows 1ms. I work remotely via VPN but my extension shows around 160ms.
I have looked at echo cancellation but can't see exactly what to do to cure this.
I have looked at the echo cancellation & training settings but they seem to be standard and I'm not sure exactly what needs tweaking.
Any pointers much appreciated.
B. Rgds
John
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How are you connected/talking to the Telrad box? Is the far end call coming to you through the Telrad box? I'm trying to understand the setup. It isn't clear from your post if the links are SIP, IAX, analogue, digital.
Kind Regards
S
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How are you connected/talking to the Telrad box? Is the far end call coming to you through the Telrad box? I'm trying to understand the setup. It isn't clear from your post if the links are SIP, IAX, analogue, digital.
The Telrad is connected to the PSTN via ISDN & DDIs etc
Asterisk is connected to Telrad via General SIP trunkline. I use a softphone as an extension on the Asterisk box.
Is that enough or do do you need more details - if so please ask !
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The echo is probably being caused by the latency placed on the call by your remote softphone. However, since your Asterisk box is only passing SIP, you cannot apply any echo cancellation because there isn't any for SIP. So unless you have EC that you can apply in your Telrad box then you have nowhere to go except to perhaps reduce the latency of your softphone. 160ms latency is a lot, even for a SIP channel. How far are you away from the box? As an example, I get less than 160 ms lag running SIP calls from UK to the USA. It may be the vpn which is causing the delay, in which case you might want to try running the softphone as a remote, rather than through the vpn, to see if it makes any difference. You night also want to hook up a phone locally to the box and see if it also causes echo.
Kind Regards
S
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The echo is probably being caused by the latency placed on the call by your remote softphone. However, since your Asterisk box is only passing SIP, you cannot apply any echo cancellation because there isn't any for SIP. So unless you have EC that you can apply in your Telrad box then you have nowhere to go except to perhaps reduce the latency of your softphone. 160ms latency is a lot, even for a SIP channel. How far are you away from the box? As an example, I get less than 160 ms lag running SIP calls from UK to the USA.
I'm in Spain and the boxes are in the UK. What sort of latency should I be trying to achieve - I know the less the better, but what's a sort of maximum recommended ?
It may be the vpn which is causing the delay, in which case you might want to try running the softphone as a remote, rather than through the vpn, to see if it makes any difference. You night also want to hook up a phone locally to the box and see if it also causes echo.
Yup, that could well make a difference I guess.
I can will try your other suggestions too & see if that makes a difference.
Would it make a difference if the Asterisk box was this end in Spain rather than in the UK ??
Thanks for the help & swift responses
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What sort of latency should I be trying to achieve
For pure SIP you can get as high as 300ms and kind-of get away with it. There will be a noticeable lag but there should be no echo. For analogue the problem is the latency across the hybrid interface (which is what causes echo). The magic number is a about 70ms. Below that the human ear doesn't hear it.
I just spoke to one of our ISP customers who has a high-availability SARK cluster just outside Alicante. He says it gives a latency of about 60ms from the UK. Now it isn't a totally fair comparison because it's sat on a 100MB fibre node but it suggests that you might be able to do better than 160.
I don't think it matters where the asterisk box is because moving Asterisk will not change the overall latency.
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Hmm. OK.
I live just outside a small village not far from Valencia and am lucky just to have broadband !
Just straightforward ping times give me around 108 ms from here to the office in the UK, and about 10ms more across the VPN and I guess it isn't going to get any better than that ! Not sure why Asterisk reports higher than that ?
Pings to say a big university site etc in the UK give times of around 80ms so there seems to be a helluva lot of time lost in the BT network..........
My wife has a Telrad VOIP handset linked via VPN back to the Telrad box which works very well. I was just experimenting with some alternatives - I don't have another handset, and run linux on my desktop so can't use the Telrad Softphone, though when I did some tests with it on another box, it pushed the line very hard and we got some drop out with it.
I have two broadband lines, but the Telrad extension really needs one exclusively due to limits on upload bandwidth so I have one dedicated to the VOIP stuff and the other to general Internet. I was hoping that with a different codec & Asterisk, I might be able to squeeze another call up the same line at the same time ! It works bandwidth wise, but clearly too high on the latency, and that won't change by moving it to the other line.
May have to go back to the drawing board............