I’m using fews SIP phone connected to WIFI and do some call testing. The quality of the voice very bad and not consistent. Lot of echo sound and voice distortion. Anyone can help me to improve the problem I’m facing? Thank you
Echo problems are due to the phones themselves. Poor quality is due to the network.
Our concern is not the phone as we had tried on testing using many type of handphone support sip client. We also had tried on different sip client. But the quality result still not satisfy and not in good quality. We would like to know that is SIP protocol causing the poor voice quality. As this had been told by some mobile manufacturer engineer. We also like to know would it run on VoWLAN would causes the poor quality and anyone had the experience how to over come it can share it hear. Thank you
I’ve run SIP WiFi phones before. The two big things when doing it are:
- They will not work well on heavily saturated wireless networks. You will get dropped calls and choppy audio.
- You almost certainly must run your APs in 802.11g mode with short preambles.
In regards to echo, it is not the SIP protocol creating the echo. If you are getting echo on many different SIP clients, you need to troubleshoot at the point where your SIP server (presumably Asterisk) connects to the telephone network. If you have echo problems there, they may not be noticeable on hardwired SIP clients but may be exaggerated by delayed packets over a heavily saturated WiFi network.
Thank you for your information. I would like to know if my entire office would like to use asterisk as the VOIP server PBX and mobile phone or SIP phone connecting via access point. What needs to be concern and any tools can use to monitor the traffic or call in terms of loss packet and call quality?
If its a reasonable phone it will have the g729 codec. I would use that - it reduces the amount of bandwidth required down to about 34 KB/s (per call 8k + headers). This is a huge amount less than say ulaw.
Wired networks, just like wireless networks can get congested, this congestion causes packet loss resulting in choppy audio and even dropped calls (when too many critical packets are lost), and echo due to high latency. Wireless networks have even more work to do considering that they have to be secured, adding even more overhead. You really are asking a lot of a basic Wireless Access Point to handle a bunch of simultaneous calls.
In addition to what jpsharp said… Echo is actually part of every call - without going into too much detail about why and how, just trust me, there is always echo on every call. however you cannot hear echo if its close enough to the source, say within about 20ms. Echo cancellation can only work on a certain range say up to 150 or 200 ms. Once the latency goes beyond this, you will hear the echo from your own voice - making the call annoying.
Which is why it is a phone problem (unless the phones are simple ones on the public network). The right place to cancel echo on VoIP networks is in the phone where the echo is being generated. For the PSTN, it should be done at the PSTN/VoIP boundary.