Echo and poor quality sound

[ebrito]

Hi;

 

I'm experiencing echo and poor quality sound on some calls and would like to know if it's normal.

Here is the scenario:

 

- Axon 1.2 installed in a Pentium D 3.4, 1GBRam, runing Windows Vista Ultimmate

- IVM 4.03 on the same machine

- Internet Connection: 800Kbps Upload x 10Mbps download

- Location: New Jesey (USA)

 

- Line used for external Calls: Broadvoice

- Line for incoming calls: DID number from DID World Wide

- Clients: Offices in Brazil and US using Express Talk

 

The results:

 

1) When calling extensions (everybody using Express Talk) sometimes we get an 1 to 1.5 second delay, sometimes a little echo.

Same thing if the two person are in Brazil or one in US and another one in Brazil.

Sometimes, the call is better than expected.

 

2) When somebody here call's a PSTN line in Brazil (here the Broadvoice line will be used), the conversation is ok.

Once in a while we get an a small delay (maximun of 1.5 second) and sometimes a little echo.

 

3) When somebody from Brazil, using a PSTN line, call the DID number, the PBX get the call and transfer to an extension also in Brasil, using Express Talk.

In this case the sound quality is always poor, sometimes the sound is choppy, sometimes cutting off, always a small (aceptable) delay, sometimes the delay is about 2 seconds.

 

Is there anything we can do to make calls from PSTN lines via DID numbers work better?

Or is it because the two person (caller and who answer) are in Brazil and the PBX i here (US)?

 

Hope somebody can help me.

 

Thanks

 

e.Brito

[pythonpoole]

- Echo

Having an echo is normal and can be fixed. You will usually get an echo if you are answering the phone with an open microphone and speakers. You should reduce the echo to nothing if you instead wear a headset. The echo is caused by your voice entering the microphone, then coming back out through the speakers and then entering the microphone again.

 

If you are still experiencing an echo problem, make sure you select the feature "Use microphone and speaker and turn echo cancellation on" in Express Talk's audio options.

 

 

- Delays

Delays are also normal, but are usually very small. An average delay for VoIP communications is probably around 0.3 - 0.5 seconds (depends on a lot of factors).

 

Large delays can be caused by:

 

- Large distance from you to VoIP provider (i.e. if your VoIP provider's SIP server is in another country, you will most definitely experience much bigger delays than using a local VoIP provider that runs from your city. Remember that the larger the distance, the larger the delay.

 

- Slow internet connection / limited bandwidth. Make sure you have a good quality internet connection with high speeds / large bandwidth. E.g. I would never recommend a 256kbps connection for people who wish to use VoIP, that is the absolute minimum. I recommend having a connection rated at around greater than 3mbit. The faster and greater bandwidth available, the smaller the delay and the less amount of quality issues in terms of cutting in and out.

 

- Having a modem or router that does not support QoS. QoS is Quality of Service and is a feature on some higher quality modems and routers that enables you to prioritize VoIP communications data over other network data to help reduce quality loss and also ensure VoIP call data is sent ahead of other less important data (e.g. regular web browsing), If you are not using a router with QoS on, I suggest you enable it, or buy a router which supports it.

 

 

- Poor Quality

 

Like delays, poor quality can also be caused by many factors. In fact, many of the same factors for delays are also factors which affect quality loss, such as:

 

- Internet connection quality / speed / bandwidth available

- Whether QoS is enabled on the routers in your network

- Distance from you and your ISP, distance to the VoIP provider

 

One other thing that might help is Express Talk's "Prefer lower bandwidth usage (But with lower audio quality)

Although the actual maximum quality of the audio is reduced when this setting is selected, much less bandwidth is used.

 

This means that the call should not experience cut outs, or connection problems where you can hear the person one second and not the next. This may be a viable option in your situation, but essentially it comes down to whether you want pure clarity and perfect quality but with voices cutting in and out, or reduced clarity, a small bit of fuzziness, but a consistent connection that doesn't keep dropping in and out.