[asterisk-users] What are the reasons for VoIP echo?

[Steve Underwood]

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
> <jsullivan at opensourcedevel.com>  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<myles at techsol.org>  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