Call quality issues
From VoIP.ms Wiki
There are different factors that can affect our calls, sound issues can be different and related to different things, we will try to mention here some suggestions, so we can identify which type of issue are we experiencing and what things we need to check to start a diagnostic by ourselves:
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Choppy/Robotic voice
Choose a sever
VoIP.ms offer different servers (13 different servers to be exact) that we can use to register with their service, a good recommendation is to send a ping to all of them or the ones that are closer to our location, this way we can verify the latency and pick the best option for us.
How to send a ping?
- Windows: Open up the DOS prompt and enter cmd /k ping xxxx.voip.ms on the box in windows (replace xxxx for the name of the server you want to ping).
- Mac: Open a "Console Aplication" and write "ping xxxx.voip.ms" (replace xxxx for the name of the server you want to ping).
If pings results are not consistent, you may have an issue with Jitter, you can work around with this issue by adjusting the "Network Jitter Level" setting on your VoIP device.
Network traffic
One of the main reasons sound issues may occur is based on the traffic or congestion on the network. First thing to try is check if the issue can be duplicated is making an internal call with the provider, for example using an Echo test application or a voicemail.
Some symptoms that can be present because of the lack of bandwidth available:
- Audio cutting in and out (choppy).
- Voice sounding robotic, like if you were talking under the water.
- Adio slowing down or speeding up, intermittently during the call.
Now, for test if the bandiwidth is affecting our calls:
- Disconnect all the devices from the network
- Disable wireless, to make sure no one else is using your internet.
- If your router has QoS, disable it.
- If you were using software to download stuff from Internet (e.g. Torrents) wait a few minutes for this traffic to subside.
After following all these suggestions, use a single device and try to make a call, if the audio quality is fine, you are probably dealing with lack of bandwidth, and for this case the use of QoS is recommended, and make sure the set up is well done.
Test codecs
Test with all the codecs such as g711u, g729 and GSM. Sometimes the issues with the audio can be related with the codecs in use, either because the codec we are using is consuming too much bandiwidth for our connection or there is a chance also the device we are using is not supporting this codec very well, or it works better with a different one. In any case, this test can also help in the diagnostic.
Check in your Account or sub account settings, which codec you are allowing, you can test allowing one by one, until you get the best result.
Check your ISP
After following these suggestions, you still experience sound issues? You may consider to contact your ISP (Internet provider) just to confirm the issue is not related with them.
Tones during calls
Another issue related with the quality during your calls, is when you can hear beep tones during a call, like if someone is pressing a button on the phone or trying to dial. This is usually known as "talk-off" and the device is interprete the voice as a DTMF digit.
Suggestions to follow:
- Upgrade the firmware in your device, sometimes these bugs are fixed in recent versions.
- Change your DTMF Tx Method to InBand (you have to change this setting in your device and in your account or sub account settings). Test if the DTMF tones are working fine, dial 4747 for this test.
- If Inband doest not work for you, test with DTMF Process INFO and DTMF Process AVT to No, if the options are available in the device.
- Another alternative is as follows: DTMF Tx Method: AVT, DTMF Tx Mode: Strict, DTMF TX Strict Hold Off time: 70.
Echo during calls
We have different factors that can cause Echo during the calls, we will review some suggestions to work with:
- Check the volume on the phone is not too loud, it is possible the phone is causing the issue.
- Make a call dialing 4443 for echo test and see if you can reproduce the same situation with this test.
- Again, check the firmware on the device, usually this can help to reduce the echo if you do not have the latest firmware.
- The default gain on some devices, is typically too high and can cause echo. Adjust the FXS Port Input Gain and FXS Port Output Gain, one at a time, in increments of three. You can test using -1 and -11.
Note: Input Gain = how you sound to the other party. Output Gain = how the other party sounds to you.
- If the above does not solve your problem, and you have a Linksys device, verify that Echo Canc Enable, Echo Canc Adapt Enable, and Echo Supp Enable are set to Yes. (These are default settings.)
- If you use laptop (integrated mic/speakers), echo can be caused by microphone catching noise from speakers. Try lowering MIC Input sensitivity.
One-Way Audio
You can hear the other party but they can not hear you, and vice-versa. When a situation like this is present, is know as "one-way audio", and usually is related with the NAT. Let's try with the following suggestions:
- From the account or sub account settings, select always NAT=Yes (is the option recommended by VoIP.ms).
- Only as a test, place the device in DMZ, to make sure if the issue is related with the NAT, even if this work, do not let the device in DMZ.
- Your router is appropiate for VoIP? If you have a router and a modem, try to bypass the router to verify if the issue gets duplicated.
- If your router includes a SIP ALG and/or SPI Firewall setting please ensure that it is disabled. That setting is common in D-Link and Netgear routers.
- If you have an ATA device (Linksys) you can work on the following settings:
Under SIP page.
- RTP Packet Size: 0.020
- G729a Codec Name: G729
- G729b Codec Name: G729
Under the Line page.
- NAT Mapping Enable: Yes
- NAT Keep Alive Enable: Yes
- Preferred Codec: G711u
- Use Pref Codec Only: No
