From: Michael S. <mi...@st...> - 2008-08-21 13:04:30
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>From my experience echo in VoIP is not caused by network latency. VoIP CODECs are required to eliminate echo from the TDM or analog side of the CODEC before creating the RTP packets. This means that there is very little residual echo in the IP domain. With very little echo left, the amount of latency in the IP network does not contribute to echo. If there are no echo cancellers in a traditional telephone network then latency multiplies the echo duration. This is why long distance carriers historically used echo cancellers where local providers generally did not need them. Once the call goes through an echo canceller (e.g. in the VoIP CODEC) then the amount of latency "after" that point does not contribute to echo. Traditionally echo comes from two primary sources; acoustic echo is generated when the sound is somehow reflected from the earpiece or speaker back into the microphone; hybrid echo is caused by the 2-4 wire conversion that occurs between an analog line and a digital telephony line (i.e. TDM circuit). In VoIP there is a third major source, bad echo canceller algorithm in the CODEC. I would suspect a bad CODEC which requires either a software update on your Sipura AT or a different TA altogether. Another possibility, although remote, is some type of cross wiring between the FSX and FXO sides of the Sipura in your home. This is not really echo but will sound like echo. If you use an analog phone to make a local call, and those wires were somehow coupled, you would hear both the primary audio and the delayed version generated after the VoIP hop. It is very easy to exclude this possibility; simply use a softclient on a PC or a pure VoIP phone to make the local call. This way the FXS port is not in use. You can also test the other port by making a call from the softclient to an analog phone connected to the FXS port. Personally, I'd try a different TA. Michael -----Original Message----- From: mis...@li... [mailto:mis...@li...] On Behalf Of Pete Flaherty Sent: Thursday, August 21, 2008 7:34 AM To: The main list for the MisterHouse home automation program; Just Subject: Re: [mh] Telephony Question Garry, Just a thought, echo is caused by latency, and if all is local latency is usually related to the network. And this is usually caused by switches in hte network. So here is what shoudll be checked: - that you are using a Switch and not a Hub for you ether - if it is a switch I've found that newer Linksys and Cisco switches seem to have the proper QoS built in, others are a crap shoot. If you can Segregate the VoIP traffic from the LAN traffic (at least for a test) . because VoIP uses UDP and there is no garuntee for delivery/transport to the other side, then You'll know if its a hardware or network thing Just my $0.02 -- -Pete Flaherty http://www.lpcomet.com http://www.mraudrey.net http://www.hauntedacrewoods.com On Tue, August 19, 2008 2:08 am, Dave Stenhouse wrote: > Garry, > If you half-tap the line on the telco side of the ATA, then you can > simultaneously ring a non-asterisk phone and some mythical phone (or > ring group) in asterisk. Then after x number of rings the call would be > answered by *'s voicemail. > Or you could try a different ATA for the FXS functionality. I have been > using a Grandstream HT-386 for a couple years now with no trouble at > all. Very high WAF. > -Dave > > Garry Doucette wrote: >> Hi Folks >> >> I'm looking for some advice for my telephone set up. >> >> Right now I have a Sipura 3000 ATA that I used to tie my PSTN line into >> my >> Asterisk server. It works very well for incoming calls on the PSTN that >> transfer through to the IVR in Asterisk. That's not a problem. The FXS >> side >> of the the ATA, however, is a bust. I just can't seem to eliminate the >> echo >> issues. I've been trying for months. I thought I had it working well but >> I >> found out the WAF was,in fact, quite low. So I've had to take down the >> ATA >> and go back to a direct connection telephone to PSTN. >> >> So, I'm looking for an alternate set up that would allow direct >> telephone to >> PSTN connections but still be able to retrieve voicemail, etc. from the >> Asterisk server. >> >> I was wondering what others are doing... >> >> Regards, >> >> Garry >> >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> ________________________________________________________ >> To unsubscribe from this list, go to: >> http://sourceforge.net/mail/?group_id=1365 >> >> >> > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's > challenge > Build the coolest Linux based applications with Moblin SDK & win great > prizes > Grand prize is a trip for two to an Open Source event anywhere in the > world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ________________________________________________________ > To unsubscribe from this list, go to: > http://sourceforge.net/mail/?group_id=1365 > > ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ ________________________________________________________ To unsubscribe from this list, go to: http://sourceforge.net/mail/?group_id=1365 |