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This is a discussion on Does anyone offer VOIP (SIP or IAX) dial-in numbers WITHOUT echocancellation within the uk.telecom.voip forums, part of the Newsgroup Forums category; All I want is a UK (and maybe also a USA) dial-in number which forwards A-law (or ulaw ...
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All I want is a UK (and maybe also a USA) dial-in number which
forwards A-law (or ulaw in USA) audio packets from the PSTN to my asterisk server without changing the audio in any way. This means no echo cancellation. Can anyone offer such a service? I could do it myself with a group of ISDN2 lines or preferably an ISDN30 connection, but this is rather expensive. In any case, why do VOIP service providers insist on echo cancelling their connections when this job is best done at the end points? John |
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In message <9fde148d-db61-4364-8813-6e93424b7037@59g2000hsb.googlegroups.com>
[email]jrwalliker@gmail.com[/email] wrote: [color=blue] > In any case, why do VOIP service providers insist on echo cancelling > their connections when this job is best done at the end points?[/color] I don't know; but here are some points to think about. VoIP shouldn't inherently have any echo. The acoustic path from earpiece to microphone is at a /very/ low level, surprisingly low, unless the incoming audio is on speakers - and in this case it would definitely be best done at the endpoint. Most echo is caused by imperfect termination of an analogue line. Any echo cancellation is easiest performed when the path delay is shortest. Look at the segments of a call, see which are analogue and which are VoIP, and then work out where the echo is best performed. Another point: acoustic paths are unpredictable and subject to variation as the speaker, microphone and other people and objects move around in a room, whereas imperfect analogue termination is constant over a very long period. Dave |
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[email]jrwalliker@gmail.com[/email] wrote:[color=blue]
> All I want is a UK (and maybe also a USA) dial-in number which > forwards A-law (or ulaw in USA) audio packets from the PSTN to my > asterisk server without changing the audio in any way. This means no > echo cancellation. > > Can anyone offer such a service?[/color] we do this -- peter gradwell. gradwell dot com Ltd. [url]http://www.gradwell.com/[/url] <email/web_hosting/business_voip> <t:01225 800 810> |
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On 24 Jul, 22:47, Peter Gradwell <pe...@gradwell.com> wrote:
[color=blue] > we do this[/color] Peter, I'm very pleased to hear it as I am already one of your customers! I emailed your support staff the other day and received the following: [color=blue] >Subject: Case Update: 16090 - [g SUPPORT] Inbound uk number without echo cancellation? > Status: Closed > Link: Click here >Response: >Hi John, > >I have checked with our UK DID numbering providers and I am afraid we cannot offer this functionality. > >We can only disable echo cancellation on an E1 circuit basis and not on a per number basis. > >Please do let us know if we can assist you any further, >Regards, > >... >[/color] If there really is a way in which you could set some of my inbound numbers up (or create new ones specially for this purpose) without e/c that would be really useful. For the benefit of those who wonder "why?" - I'm doing automated testing of mobile handsfree echo cancellers using test software on an asterisk server. Having extra cancellers in the way just messes things up. Regards, John |
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On 24 Jul, 20:18, Dave Higton <davehig...@dsl.pipex.com> wrote:[color=blue]
> VoIP shouldn't inherently have any echo. *The acoustic path from > earpiece to microphone is at a /very/ low level, surprisingly > low, unless the incoming audio is on speakers - and in this case[/color] With a standard BT badged phone on an "analogue" line I typicall see an echo return loss of about -25dB. John |
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In message <6115e09e-b958-42e6-9178-f1e342ac4e34@p31g2000prf.googlegroups.com>
[email]jrwalliker@gmail.com[/email] wrote: [color=blue] > On 24 Jul, 20:18, Dave Higton <davehig...@dsl.pipex.com> wrote:[color=green] > > VoIP shouldn't inherently have any echo. *The acoustic path from > > earpiece to microphone is at a /very/ low level, surprisingly > > low, unless the incoming audio is on speakers - and in this case[/color] > > With a standard BT badged phone on an "analogue" line I typicall see > an echo return loss of about -25dB.[/color] Sure, but my point is that that's caused by imperfect termination of the analogue line, and not by acoustic feedthrough from earpiece to microphone. Dave |
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> > With a standard BT badged phone on an "analogue" line I typicall see[color=blue][color=green]
> > an echo return loss of about -25dB.[/color] > > Sure, but my point is that that's caused by imperfect termination > of the analogue line, and not by acoustic feedthrough from earpiece > to microphone. > > Dave[/color] That will depend a lot on the particular phone and line, but my experience has been that the echo is often a mixture of mismatch reflection and acoustic coupling. It is certainly interesting to experiment with the source impedance settings on voip - analogue adapters. An incorrect setting here can easily dominate any acoustic echo. I still wonder why VOIP suppliers seem so keen on echo cancellation, when the end points are normally VOIP phones or computers which do the echo cancellation themselves. Why spend money on a process which is probably not necessary and which if overdone can degrade the signal? John |
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On Sun, 27 Jul 2008 08:49:27 -0700, jrwalliker wrote:
[color=blue] > I still wonder why VOIP suppliers seem so keen on echo cancellation, > when the end points are normally VOIP phones or computers which do the > echo cancellation themselves. Why spend money on a process which is > probably not necessary and which if overdone can degrade the signal?[/color] Perhaps the cost of implementing it across the board is less than the cost of fielding support queries from people who do have echo problems. Perhaps all the ITSPs you've tested use voice platforms that don't allow echo cancellation settings to be set per-extension. -- <http://ale.cx/> (AIM:troffasky) (UnSoEsNpEaTm@ale.cx) 19:42:25 up 15 days, 22:18, 3 users, load average: 0.08, 0.06, 0.01 Convergence, n: The act of using separate DSL circuits for voice and data |
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