On 10/02/2009 09:18 AM, Martin wrote: > Are you saying there are half duplex phones out there ???? with half > duplex speakerphones ? > Practically all analogue speakerphones are half duplex. Only a small number of analogue phones ever implemented a proper echo canceller based speakerphone - usually ones which included the necessary DSP power for other purposes, like answering machine functions. > All analog phones are full duplex ... > > Anyways the echo can be created by the analog phone even when it's > connected to the > sip ata or even the sip phone ... then you usually have acoustic echo > which goes from speaker > to microphone of the handset ... that should be cancelled by the sip > phone/device... or someone out there will > hear echo > > Martin > > On Thu, Oct 1, 2009 at 7:57 PM, John A. Sullivan III > <[email protected]> wrote: > >> I'm quite new to all this but I was under the impression that most >> electrically induced echo was at the physical interface to the PSTN. If >> one is using SIP trunking, I would think this would point to a carrier >> issue. >> >> We also hit an interesting problem with echo today but I don't think >> this is the issue Myles is having. We installed fairly high end phones >> with full duplex speakerphones. Callers are having a bad problem with >> echo when the users use the speakerphone. Because it is full duplex >> rather than half, if the speakerphone volume and speakerphone mike >> volume are turned up, the callers are indeed hearing themselves by >> virtue of the higher quality full duplex! >> >> On Thu, 2009-10-01 at 19:36 -0500, Martin wrote: >> >>> if a user calling you hears echo of himself then it's the fault of >>> your sip device/sip phone. >>> The manufacturer must be using a cheap or an open source echo canceller ... >>> >>> try getting a different sip device made by some 'normal' company like >>> polycom or linksys/cisco >>> >>> Martin >>> >>> On Thu, Oct 1, 2009 at 9:10 AM, Myles Wakeham<[email protected]> wrote: >>> >>>> I have an Asterisk 1.4.2 system that has been installed for about 3 >>>> months now in our home. We converted all of our phones to SIP phones, >>>> and use two different trunk providers (BroadVoice for incoming& >>>> FlowRoute for outgoing). >>>> >>>> Most of the time its working flawlessly. But about 1/3rd of the calls >>>> that come into us complain of an echo and what is best described as >>>> latency issues. Its not consistent though. I was on the phone with an >>>> insurance company yesterday for about 1 hour and the call was perfect (I >>>> originated the call which used Flowroute for the SIP provider). >>>> >>>> What seems to be a pattern here is cell phones. When we receive a call >>>> from a cell phone, or from certain people on certain phone systems, they >>>> consistently complain of echo in the call. Its far less regular when we >>>> originate the call, which suggested to me that the problem might be with >>>> Broadvoice. But I'm now hearing that us calling back the party doesn't >>>> always solve the problem either. >>>> >>>> We upgraded our Internet feed (we're on a cable Internet through our >>>> cable company, with 12mb/s down, 1.5mb/s up) and that seems to have >>>> helped but not solved this problem. From what I can see, its some form >>>> of latency issue. We use IPCop as a firewall for our Internet access, >>>> but have turned off any IDS on it so that its running fast. I can play >>>> online computer games through the network with no issues at all, so I >>>> don't think its slowing down the traffic and if it was I'd expect this >>>> problem to be occurring consistently on all calls. >>>> >>>> Are there any tweaks that I can do with Asterisk to increase the network >>>> performance to reduce these issues? Have others who have experienced >>>> this been able to identify the issues to external VoIP SIP providers >>>> only, or does our system have something to do with all of this? At the >>>> time of the calls coming in, IPCop is telling me that we don't have more >>>> than 100K/s of bandwidth in use, and according to the network bandwidth >>>> graphs there, even with 2 people on the phone at the same time, the >>>> bandwidth never seems to exceed 300K/s, so I think we have plenty of >>>> headroom for this. I checked with our cable provider for issues with >>>> modem latency, and they couldn't detect anything. Again, I'm not >>>> experiencing any lag issues with computer games, particularly those that >>>> are heavy in interactivity, so I don't think that is the reason. >>>> >>>> Any suggestions as to what could be tweaked would be greatly appreciated. >>>> Steve
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
