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Re: [mh] Telephony Question SIPURA SPA3102 ?
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From: Tom M. <tma...@sa...> - 2008-08-28 13:56:26
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I use the SPA3102 and I have no problems with echo. I do have issues getting DISA (Direct Inward System Access, like having an internal extension as an external caller) because it doesn't seem to detect my DTMF tones properly. -tom m. _____ From: mis...@li... [mailto:mis...@li...] On Behalf Of Gianni Veloce Sent: Thursday, August 28, 2008 9:51 AM To: The main list for the MisterHouse home automation program Subject: Re: [mh] Telephony Question SIPURA SPA3102 ? Hi as I am intersted on buying an ATA, if SIPURA SPA3102 is not recommended, could someone recommend other better choices (at the same price level) thank you GV ----- Original Message ---- From: Andy M. <myi...@st...> To: gar...@sh...; The main list for the MisterHouse home automation program <mis...@li...> Sent: Friday, August 22, 2008 7:48:32 AM Subject: Re: [mh] Telephony Question Garry, Have you seen the links below? Admittedly they refer to the Sipura SPA3102 - which I have and don't recommend any more - but they are part of the same series and "probably" share the same design flaw...... http://www.geekzone.co.nz/forums.asp?ForumId=43 <http://www.geekzone.co.nz/forums.asp?ForumId=43&TopicId=12363%5C> &TopicId=12363 http://forums.whirlpool.net.au/forum-replies-archive.cfm/717482.html Also, Linksys themselves have a help article on reducing echo in this series. http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_fa qid=5167 <http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_f aqid=5167&p_created=1168646951&p_sid=II4Twyyi&p_accessibility=0&p_lva=5167&p _sp=cF9zcmNoPTEmcF9zb3J0X2J5PSZwX2dyaWRzb3J0PSZwX3Jvd19jbnQ9NyZwX3Byb2RzPSZw X2NhdHM9MTcyMSwxNzIzJnBfcHY9MS4yNDMmcF9jdj0yLjE3MjMmcF9zY2ZfbGFuZz0xJnBfcGFn ZT0xJnBfc2VhcmNoX3RleHQ9ZWNobw**&p_li=&p_topview=1> &p_created=1168646951&p_sid=II4Twyyi&p_accessibility=0&p_lva=5167&p_sp=cF9zc mNoPTEmcF9zb3J0X2J5PSZwX2dyaWRzb3J0PSZwX3Jvd19jbnQ9NyZwX3Byb2RzPSZwX2NhdHM9M TcyMSwxNzIzJnBfcHY9MS4yNDMmcF9jdj0yLjE3MjMmcF9zY2ZfbGFuZz0xJnBfcGFnZT0xJnBfc 2VhcmNoX3RleHQ9ZWNobw**&p_li=&p_topview=1 Andy. Garry Doucette wrote: Pete, Michael Thanks very much for your insight. >From what I understand, the issue with the Sipura 3000 is that the PSTN call is converted from analog to digital and then a SIP call is established between the FXO and FXS ports in the device. It's this conversion process that causes the latency and hence the echo. There's quite a few settings in the device that deal with echo, but after months of playing with them I just couldn't get a satisfactory result. It would be nice is if the FXS side of the ATA would simply make a direct connection to FXO when you dial a PSTN call (the SPA3000 does this if you unplug it) and would connect to the network if you want to make a VOIP call. I think I will research another ATA, maybe one of the Grandstream ones... Thanks again. Garry -----Original Message----- From: mis...@li... [mailto:mis...@li...] On Behalf Of Michael Stovenour Sent: Thursday, August 21, 2008 6:04 AM To: 'The main list for the MisterHouse home automation program' Subject: Re: [mh] Telephony Question >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.lpcomet.com/> http://www.mraudrey.net <http://www.mraudrey.net/> http://www.hauntedacrewoods.com <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 <http://moblin-contest.org/redirect.php?banner_id=100&url=/> &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 <http://moblin-contest.org/redirect.php?banner_id=100&url=/> &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 <http://moblin-contest.org/redirect.php?banner_id=100&url=/> &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 <http://moblin-contest.org/redirect.php?banner_id=100&url=/> &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 <http://moblin-contest.org/redirect.php?banner_id=100&url=/> &url=/ ________________________________________________________ To unsubscribe from this list, go to: http://sourceforge.net/mail/?group_id=1365 |