Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS Can i control the cancellation with the zapata.conf ? I have tried this with "echocancel=..." and so on, no luck :( I would be glad to get some help, the Docs of Asterisk dont explain how to cancel Echos in ! SIP !
On Thu, 2006-10-26 at 12:18 +0200, Stefan Agethen wrote:> Hi, > > i am from Germany, so excuse my School English. > > I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update > of Asterisk 2 wooks ago, Echos accure in my SIP Calls. > > I use SNOM 360, sometimes there is no echo (for example if i call myself > via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) > but if i call other people there occures Echo many times. The Routing is > always the same : > SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS > > Can i control the cancellation with the zapata.conf ?The snom phones are pretty decent devices and shouldn't introduce echo. Your latency might be too high between asterisk + voipprovider introducing a delay that is noticed as echo. You are hearing the echo that is introduced on the callers side. As I understand it, when you are calling someone there is no zap involved and thus you can't cancel it with zapata.conf. if you look at voip-info.org [1] you'll find a good explanation why you can't use an echo canceller to cancel that sort of echo. So, check the path between you and the voipprovider, e.g. connection saturation, ping times etc (This is assuming you have a proper lan connection between asterisk/snom) Conrad [1] http://www.voip-info.org/wiki/index.php?page=Asterisk+Echo +Cancellation
On Thu, Oct 26, 2006 at 12:18:20PM +0200, Stefan Agethen wrote:> Hi, > > i am from Germany, so excuse my School English. > > I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update > of Asterisk 2 wooks ago, Echos accure in my SIP Calls. > > I use SNOM 360, sometimes there is no echo (for example if i call myself > via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) > but if i call other people there occures Echo many times. The Routing is > always the same : > SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS > > Can i control the cancellation with the zapata.conf ?Sure, but only for Zaptel channels. Not for mISDN channels. If you use ZapBRI, this would be the place to configure echo cancelling. -- Tzafrir Cohen icq#16849755 jabber:tzafrir@jabber.org +972-50-7952406 mailto:tzafrir.cohen@xorcom.com http://www.xorcom.com iax:guest@local.xorcom.com/tzafrir
>> Hi, >> >> i am from Germany, so excuse my School English. >> >> I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update >> of Asterisk 2 wooks ago, Echos accure in my SIP Calls. >> >> I use SNOM 360, sometimes there is no echo (for example if i call myself >> via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) >> but if i call other people there occures Echo many times. The Routing is >> always the same : >> SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS >> >> Can i control the cancellation with the zapata.conf ? >> > > The snom phones are pretty decent devices and shouldn't introduce echo. > Your latency might be too high between asterisk + voipprovider > introducing a delay that is noticed as echo. > You are hearing the echo that is introduced on the callers side. > As I understand it, when you are calling someone there is no zap > involved and thus you can't cancel it with zapata.conf. > if you look at voip-info.org [1] you'll find a good explanation why you > can't use an echo canceller to cancel that sort of echo. > So, check the path between you and the voipprovider, e.g. connection > saturation, ping times etc > (This is assuming you have a proper lan connection between > asterisk/snom) > > Conrad > > [1] http://www.voip-info.org/wiki/index.php?page=Asterisk+Echo > +Cancellation >Hi Conrad, thanks for your help, this is the way i understand it all the time, a year ago i have optimized my Business Lan for VoIP and there is no loss or lag anymore, the Provider seems to be okay, i have pinged him for one day with MTR, no great loss or high ping. Some days ago i have read a Thread about EC-Cancellation in SIP Calls with the zapata.conf and never understood how this could work, thats the beginning of my question ;) In my case, there is no Zap, you are right. So i must start at the beginning and search for a lag... The EC started two or weeks ago after one year of great communication, in this time i updated Asterisk from 1.2.10 to 1.2.12.1 and zaptel from 1.2.7 to 1.2.9 ... I will watch the Quality and the latency next time....Thx for your time ! Stefan
On 2006-10-26 03:18:20 -0700, Stefan Agethen <stagethen@baeckereiagethen.de> said:> Hi, > > i am from Germany, so excuse my School English. > > I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update > of Asterisk 2 wooks ago, Echos accure in my SIP Calls. > > I use SNOM 360, sometimes there is no echo (for example if i call > myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) > but if i call other people there occures Echo many times. The Routing > is always the same : > SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS > > Can i control the cancellation with the zapata.conf ? > > I have tried this with "echocancel=..." and so on, no luck :( > > I would be glad to get some help, the Docs of Asterisk dont explain how > to cancel Echos in ! SIP !Are you hearing the echo, or is the far end party?
Echo is generated by the analog end to where you place the call, not the IP side of it. As far as I know the echo cancelation in the Asterisk can only be tweaked in the zapata.conf (since IP calls don't generate it) I'm afraid there is little you can do to here. Alyed ---------------------------------------- Return-Path: <asterisk-users-bounces@lists.digium.com> Thu Oct 26 13:27:22 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Thu, 26 Oct 2006 13:27:22 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) On 2006-10-26 03:18:20 -0700, Stefan Agethen said:> Hi, > > i am from Germany, so excuse my School English. > > I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update > of Asterisk 2 wooks ago, Echos accure in my SIP Calls. > > I use SNOM 360, sometimes there is no echo (for example if i call > myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) > but if i call other people there occures Echo many times. The Routing > is always the same : > SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS > > Can i control the cancellation with the zapata.conf ? > > I have tried this with "echocancel=..." and so on, no luck :( > > I would be glad to get some help, the Docs of Asterisk dont explain how > to cancel Echos in ! SIP !Are you hearing the echo, or is the far end party? _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061026/0143ee7d/attachment.htm
I am surprised that you are getting echo on SIP calls. You can get echo in two scenarios on SIP calls. 1. If SIP calls are crossing to PSTN (inbound/outbound). Here you need to enable echo canceller and AGGRESSIVE if needed in zconfig.h. 2. Second source of echo on SIP calls could be ACOUSTIC. The phone sets you are using may not handle this well. In my experience sound quality deteriorates if there is network trouble or congestion on SIP calls I hope this helps. Michael
You will also perceive jitter as echo If any links are getting busy and routers or switches have to buffer you will hear what sounds like echo, not to mention if you have a high packet loss also Of course jitter would have to be above 100ms or so to be noticeable as far as acoustic echo, i have had to put 192ms tail ec on pri direct from carrier because of so many networks interconnecting and doing poor jobs and that 192ms is not going to be enough shortly yes traditionally telco echo originated at 2wire to 4 wire transition points or on hybrids hence it is usually referred to as hybrid echo versus acoustic echo which does happen in an all digital call. This is one thing the better quality phones give you some control of. I am starting to look at dedicated aec hardware to handle even all IP calls On Oct 26, 2006, at 9:56 PM, Michael Araba wrote:> I am surprised that you are getting echo on SIP calls. You can get > echo > in two scenarios on SIP calls. > > 1. If SIP calls are crossing to PSTN (inbound/outbound). Here you need > to enable echo canceller and AGGRESSIVE if needed in zconfig.h. > > 2. Second source of echo on SIP calls could be ACOUSTIC. The phone > sets > you are using may not handle this well. > > In my experience sound quality deteriorates if there is network > trouble > or congestion on SIP calls > > I hope this helps. > > Michael > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
>> Hi, >> >> i am from Germany, so excuse my School English. >> >> I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my >> update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. >> >> I use SNOM 360, sometimes there is no echo (for example if i call >> myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) >> but if i call other people there occures Echo many times. The Routing >> is always the same : >> SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS >> >> Can i control the cancellation with the zapata.conf ? >> >> I have tried this with "echocancel=..." and so on, no luck :( >> >> I would be glad to get some help, the Docs of Asterisk dont explain >> how to cancel Echos in ! SIP ! > > > Are you hearing the echo, or is the far end party? > >I can hear the Echo, the end party never got this problem.
On 2006-10-26 23:02:40 -0700, Stefan Agethen <stagethen@baeckereiagethen.de> said:> >>> Hi, >>> >>> i am from Germany, so excuse my School English. >>> >>> I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update >>> of Asterisk 2 wooks ago, Echos accure in my SIP Calls. >>> >>> I use SNOM 360, sometimes there is no echo (for example if i call >>> myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) >>> but if i call other people there occures Echo many times. The Routing >>> is always the same : >>> SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS >>> >>> Can i control the cancellation with the zapata.conf ? >>> >>> I have tried this with "echocancel=..." and so on, no luck :( >>> >>> I would be glad to get some help, the Docs of Asterisk dont explain how >>> to cancel Echos in ! SIP ! >> >> >> Are you hearing the echo, or is the far end party? >> >> > I can hear the Echo, the end party never got this problem.If you can adjust the microphone gain on your local handset down, you may find that reduces audio coming back from the far ends mic (ie If you audio is very loud it could be bleeding across from the far end ear piece to the far end mic. Worth a try anyhow... Marty
Message: 7> Date: Thu, 26 Oct 2006 22:56:58 -0400 > From: "Michael Araba" <maraba@quikcat.com> > Subject: [asterisk-users] RE: ECHO Cancellation in SIP Calls > To: <asterisk-users@lists.digium.com> > Message-ID: > <8C18AA7CE9A6804090E8235C22C3BBAF495356@BE02.exg3.exghost.com> > Content-Type: text/plain; charset="us-ascii" > > I am surprised that you are getting echo on SIP calls. You can get echo > in two scenarios on SIP calls. > > 1. If SIP calls are crossing to PSTN (inbound/outbound). Here you need > to enable echo canceller and AGGRESSIVE if needed in zconfig.h. > > 2. Second source of echo on SIP calls could be ACOUSTIC. The phone sets > you are using may not handle this well. > > In my experience sound quality deteriorates if there is network trouble > or congestion on SIP calls > > I hope this helps. > > Michael > >Hi Michael, For sure, i can get echo in the 2 to 4 wire scenario, this is right, but this cant be happen in MY way, only the provider can produce this scenario, my asterisk use zap and isdn, but the echo occure in pure sip calls, in my zap and isdn channels i use the patch from mgernoth, named "mg2", great stuff. The second is one echo i already know, one other caller parties use very cheap phones, so the sound of the telephone speaker is not shielded enough to put no sound in the telephone mic - this is not the case with my phones, i use SNOM, they are build to used with VoIP and the best one i know, in my case. I checked the latency and loss between me and my provider this morning again, and i figured out a routing point which lost 3% of my packets, first time for me to see this after one year of working good, i wrote a mail to my provider, and asked him to check this on his own, but i cant imagine that this produce all the echo...must wait, i guess. I tested my Network, good results, tested other VoIP Provider's Server, Result is good to ok. Recap : To minimize echo i can check : Phone (ok), Channels in Asterisk (crossing) (ok), My Network Connections between Phone and Asterisk (ok), Network between Asterisk and Router (ok), Connection, Loss and Latency between Asterisk/Router and my VoiPProvider (waiting..) Any other ways to produce echo in pure *SIP* ! Thanks for your great help ! Stefan