Hi Scott Thank you very much.
Indeed on a recording of conversations, with a standalone gateway or a card in an asterisk server, I can see a huge amplitude imbalance between rx/tx on most installation here in Belgium. In this one the amplitude difference is even greater. Iirc in the past we once lowered the mic/tx volume on snom phones for that reason (these customers were used to talk loud in an open space, and had lots of echo problems). Maybe the Polycom phones we used in this installation (the HD IP650 models) have a louder tx volume than the average sip phones and so on this installation we are really,really too loud ? Or maybe there is a gain that might have been tuned on the telco side for these specific lines in the past ? Anyway, yesterday, I asked my colleague to decrease the tx gain on the patton (instead of increasing rx gain as he tried). This should have a similar effect as adjusting the phone tx gain I guess (I prefer this because they also have simboxes). Really glad to have a confirmation on your side and I hope this will fix the problem. If this doesn't work we'll try lowering the volume on the polycom instead. In the past I tried to find a telco test number (milliwatt ?) to adjust tx/rx gains to their recommended values, but it seems there is none (or at least, none public) in Belgium :( Any Idea how I could get a good value for tx/rx with BRI lines (lots of info on the net, but specific to asterisk, and mostly for analog zaptel cards, as BRI are not popular ouside Europe) ? Should I try and align the average tx amplitude on the average rx I currently get (this would lead to a huge decrease of tx to match rx however as far as I can see in the recordings) ? I'm already under the impression rx is also too low, the customers always increase their handset volume on the phones. Thanks again Gaetan On 09/09/2009, at 16:14, Scott Richesson wrote: > This may be obvious, but I've solved the far-end echo problems at my > company by decreasing the transmit volume on my Polycom phones. > > Polycom does not recommend this, but it did work. In etc/sipxpbx/ > polycom/polycom_sip.cfg, change voice.gain.tx.analog.handset to 0 > (from 6). > > The transmit volume is still plenty loud over our PRI. > > Scott > > -----Original Message----- > From: Gaëtan Minet [mailto:gminet...@mcit.be] > Sent: Wednesday, September 09, 2009 5:41 AM > To: Paweł Pierścionek; sipx-users@list.sipfoundry.org > Subject: Re: [sipx-users] Fwd: Patton smartnode 463x : is 25ms > echocancel tai > > Hi > > Thank you very much for your feedback. > Do you mean the EC is not active at all (even the low 25ms) for the > far-end side when we use the gateway in TE mode to connect a local sip > system to the pstn using BRAs ? > Damned, I must have missed this in the datasheet :( > > So you're not using the gateway to connect to pstn in your case, only > to connect local isdn terminals ? > > Here we get plenty of far-end echo on our last installation when using > the gateway to connect to the pstn. > We just tried the patton gateways instead of pci cards (I didn't > notice EC was only 25ms...). > Before that, we used digium b410p with 64ms HW EC on asterisk, but we > noticed that, although the echo seems to disappear, there still > remains a "crispy" echo than can be easily heard on good sip phones > like polycom... thank you HD audio :) . We are still waiting for an > answer from Digium support. > > One or two year ago we almost never heard echo, but now we have more > and more, mainly on companies that receive calls from residential > users. > I suspect this could be due to the increasing popularity of "triple > play" offers here where the providers probably use cheap ATAs, and the > end users usually have cheap analog home phones. I guess the latency > and round trip can easily go above 64ms on these setups. > > Are you aware of good gateways with 128ms echo canceller (Audiocodes > maybe ?). These are probably way more expensive but we could at least > try it to confirm the problem. > Maybe we should also try to build a standalone asterisk gateway with > better cards that the digium B410p then (sangoma cards have 128ms EC, > but maybe a software EC like oslec will provide better results than > the patton anyway). > > > Gaetan > > > > On 08/09/2009, at 23:44, Paweł Pierścionek wrote: > >> Hi, >> >> Remember that the EC chip does cancellation only for the echo that >> You >> cause not the far end echo. >> So if all You have are digital (ISDN) phones, no PBX + lengthly >> cables >> or multiple A/D D/A conversion nor strange SIP inside than 25ms is >> OK. >> 25ms is not ok if You use cheap SIP phones (with no EC) in LAN >> conditions where jitterbuffer is above 25ms. >> >> To sum it up - Patton 463X + all digital setup (good SIP or ISDN >> terminals) = no echo towards PSTN >> Patton 463X + cheap PBX with analog phones or analog SIP ATAs and >> mismatched impedance = possible echo >> >> Patton 463X are popular here in Poland and so far no echo problems. >> >> Pawel, >>> I'm not sure if patton actually stock/sells bra sn's in the US >>> though. >>> -----Original Message----- >>> From: "Kristian D. Guntzelman" >>> <kristian.guntzel...@realworldtime.com> >>> Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> >>> >>> Sent: 9/8/2009 4:12:41 PM >>> Subject: Re: [sipx-users] Fwd: Patton smartnode 463x : is 25ms >>> echocancel tailenough ? >>> >>> ISDN-BRI?!? Sure we do, just not very much these days. It was >>> known as the "poor man's PRI" back when PRIs were ungodly expensive. >>> >>> Supports 2b+d and was used primarily for Polycom Video conferencing >>> and data circuits. Comes in two variants. BRI-U (2-wire) and BRI-S/ >>> T (4-wire) >>> >>> - >>> Kris Guntzelman >>> >>> Sent from my iPhone >>> >>> On Sep 8, 2009, at 2:24 PM, "Picher, Michael" <mpic...@cmctechgroup.com >>> <mailto:mpic...@cmctechgroup.com>> wrote: >>> >>> We don’t have BRI here in the US… >>> >>> From: >>> sipx-users-boun...@list.sipfoundry.org<mailto:sipx-users-boun...@list.sipfoundry.org >>>> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of >>> Gaëtan Minet >>> Sent: Tuesday, September 08, 2009 11:15 AM >>> To: <mailto:sipx-users@list.sipfoundry.org> sipx-users@list.sipfoundry.org >>> <mailto:sipx-users@list.sipfoundry.org> >>> Subject: [sipx-users] Fwd: Patton smartnode 463x : is 25ms echo >>> cancel tailenough ? >>> >>> Hi, >>> >>> Nobody is using these ? >>> >>> Gaetan >>> >>> >>> Begin forwarded message: >>> >>> >>> From: Gaëtan Minet >>> <<mailto:gminet...@mcit.be>gminet...@mcit.be<mailto:gminet...@mcit. >>> b >>> e>> >>> Date: Thu 27 Aug 2009 14:22:28 GMT+02:00 >>> To: <mailto:sipx-users@list.sipfoundry.org> sipx-users@list.sipfoundry.org >>> <mailto:sipx-users@list.sipfoundry.org> >>> Subject: [sipx-users] Patton smartnode 463x : is 25ms echo cancel >>> tail enough ? >>> >>> Hi, >>> >>> Are some of you using Patton smartnode BRI gateways ? I see the >>> hardware EC is only 25ms (PRI one have 128ms). >>> Is this tail long enough for PSTN connection under regular usage ? >>> >>> Thanks for your feedback >>> >>> regards, >>> >>> Gaetan >>> >> >> _______________________________________________ >> sipx-users mailing list sipx-users@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-users >> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >> sipXecs IP PBX -- http://www.sipfoundry.org/ > > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
