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About Gerry Magill
Expertise
I am a Software Architect employed by a large multi-national communications company providing VoIP and tradtional TDM communications to Enterprise customers.

Experience
<BR><B>Past/Present clients</B><BR>IBM, SBS, Siemens, KPMG, Bank Berlin, Commerzbank<BR>
 
   

You are here:  Experts > Internet/Online > Internet Conferencing > IP Telephony > voip

Topic: IP Telephony



Expert: Gerry Magill
Date: 5/8/2008
Subject: voip

Question
QUESTION: I have a cisco voip system.  My users complain occasionally about "echo"  I've experienced it myself but its not a standard echo in the sense of like the grand canyon. its a single, delayed, clear, non reverberating bounceback of your own voice.  there's no continuing echo where it repeats over and over. I've tried going online to find sound clips of voip echo and they dont sound like mine, they're the "grand canyon" type of echo. I've run QOS reports and all looks fine.  we have a PRI going into a voice gateway, then into the network.  I can provide many more details if you want.
thanks

ANSWER: First, are you sure it is echo and not speech clipping due to varying delay in your network? Clipping will sometimes be reported as echo by users. Look for some wav file examples of that and see if its more like what you are hearing.

My experience with echo in VoIP networks is that echo is not due to the VoIP design itself, rather it is somewhere where the digital VoIP world meets analog wiring or circuitry.

The most common place to look for this is to ask the end users if they are using head-sets. These are commonly purchased after-market items bought privately by the employees themselves to make their life easier. These devices are analog in nature. They attach to the phone in some way, either by inserting themselves in the wiring of the handset or by sticking to the side of the phone in the form of a microphone/speaker.

These devices work great on traditional phones but are often sources of impedance mis-match on VoIP phones. They also have poorly shielded wiring.

First ask the users if they have such a headset. They might deny it on the phone, so a walk-by is best in my experience. If they have headsets, ask them to remove them and try again. If that was the source of the echo you have your problem. A different brand of headset may solve the issue. Not all are of the same quality.

Also, note that the headset may be on the other parties VoIP phone at the far end. Their own voice is getting echoed back from the far end by that persons headset. If the round-trip delay is above 150ms for a ping, then the user will hear that echo.

Delay is also a source of echo in itself. Most VoIP phones have built in echo cancellers or the gateways do. These echo cancellers do a great job if the window of echo reception is < 150ms. Above that, then their buffers are not large enough to capture the echo and remove it. In VoIP networks, it can be common to have round-trip delays of over 150ms - especially on WAN connections that may not have a guaranteed end-to-end QoS in place by the WAN carrier for voice traffic.

In any voice call, a user actually hears their own voice in their headset, but this is ignored by the brain if the delay is 100-150ms. Your brain has to do this or it would get confused by your own speech resonating inside your head. Once the delay gets too large, your brain cant handle that and this is what is perceived as echo.

Work out what your round trip ping times are at peak times of network traffic or at the times that users complain. You may be surprised at what you find. Many companies have USA to Europe round trip times of 300ms. Fine for data - terrible for voice.

Also check if your users are still using analog phones connected to analog to SIP gateways. Impedance mis-match here can normally be adjusted in the gateway to compensate for wiring.

Let me know if any of the above fit your scenario.

---------- FOLLOW-UP ----------

QUESTION: thanks, that's a ton of great information, but I dont think much of it applies.  for starters I absolutly know that the prime complainers arent using headsets.  2nd as far as digital meeting anolog, i'm using a PRI  so isnt that digital to digital?  as far as pings go, all I can ping is my gateway and that's coming in at =1 ms or <1ms
so that's not the problem.  like I explained, its a Single, clear bounce back of your own voice, one time.  I'll have to try and get a recording of it somehow.
thanks!

ANSWER: So the users are complaining about calls to the gateway in particular? They dont have issues calling oneanother or calling another gateway?

Are they using speaker phones or does the echo exist in the handset mode of operation?

What phones are they using? Also Cisco devices?

What codecs are you using? G711? Are echo cancellers turned on?

The PRI trunk is digital so it's not that. If it were are T1 E&M then that's something else...

Are you using SIP, H.323 or Skinny?

Do you proxy media or is it a direct connection from the phone to the gateway?

Regards,
Gerry.

---------- FOLLOW-UP ----------

QUESTION: cool, ok thats more like the line of troubleshooting I was expecting. thanks!
There is NEVER ANY echo in house, its only on inbound or outbound calls.  they aren't using speaker.  its not on every call and it seems to effect some people more than others.  I have a 7940 and talk ALOT and almost never get it.  there are 2 ladys that have 7961 phones and say they get it about 1/3 of all calls. the person on the oustide doesnt hear any problem at all, only the person here, on the voip phone.  
so, we only have 1 gateway,  echo covereage is enabled to 32 ms.  input gain is at -3 and output attenuation is +3.
looks like I'm using G.711 at 384kbps here.  I have more than one regon with remote campuses on T1s and most of those are set to G.729.  the remote campuses have their own SRST routers and use pots at those locations so only extension to extension calls flow over the Ts.
All users without execption use cisco IP phones, most are 7940s with a few 7960s here and there.
to further complicate things, the gateway does has a CAS t1 E&M coming off it to an old PBX, about 1/2 the users are on the PBX and half are on voip.  users from pbx and voip experience echo at times, indicating the problem is in the gateway.  voip to voip NEVER have echo (these calls dont flow through the gateway)  pbx to pbx dont have echo (same thing).  but both do have echo at times when calling out or calling eachother.  again, pointing to the gateway as the problem. the gateway is MGCP, no sip or skinny. gateway cpu utilization is always under 10%. no proxy, but the gateway does sit on a different subnet/vlan than the phones, but the phones are on their own subnet away from the computers, wireless etc.  there's a cisco 6513 acting as a multilayer switch to route between those vlans.  all switches are 3560 POE switches with auto-QOS enabled on all ports and each switch has its own gig uplink to the core with NO daisychaining of switches. as stated, ping times from phone vlan to gateway, like I said are <1ms
thanks for all your help!

Answer
OK - so what you need to start doing is asking the people who complain to write down the exact times and telephone numbers they were connected to when they experienced the echo. My guess is it was only on calls through the gateway to the PBX users. This is where your analog world is hitting your digital world and there is almost certainly an impedance mis-match there causing the echo.

You say the gateway is using MGCP. Is this a platform using a cisco IOS or is it another OEM vendor make / model? If it is IOS based, why not change it to SIP? Are your subscribers using SIP as their VoIP protocol? The reason I ask is because its not clear to me what is doing the protocol conversion from the end devices to the MGCP protocol or are the IP phones all MGP endpoints? Do you have a cisco callmanager or what is your VoIP proxy doing the subscriber to IP-Address location?

The T1 E&M is an analog protocol - not digital in the sence of a PRI. So there will be echo cancellers on it to setup on both ends of that link and an impedance matching needs to be done. On the TDM end of that E&M link, the telephony guys will know how to adjust that. What kind of TDM switch is it? Also, that link should be using G.711 too as the analog TDM codec to avoid transcoding in the gateway from one codec to another. Echo cancellers on the TDM / VoIP gateway can sometimes collide as they use different algorithms to do cancellation. Sometimes turning off echo cancellers on the gateway and TDM side can actually help.

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