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Is there a quick way how to optimize Wi-Fi AP/router settings for better VoIP performance? I use SIP protocol. Can I tweak RTS or fragmentation threshold values for better voice performance?

I am asking because if I use SIP on my desktop (Linphone) over Ethernet, it seems to have better voice quality then when using CIP on Android over WiFi. My current AP has pretty much the default values set for wireless:

  • RTS: 2346 (=max)
  • Fragmentation threshold: 2346 (=max)

The AP has only these "advanced" wireless settings but I can buy a better new one in case it would help for voice.

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  • Did any answer help you? if so, you should accept the answer so that the question doesn't keep popping up forever, looking for an answer. Alternatively, you could provide and accept your own answer. – Ron Maupin Aug 12 '17 at 20:08
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When you design a wireless solution for VoIP, several things must be taken into consideration:

  • Jitter: Variations in delay of packet delivery , Jitter buffers (used to compensate for varying delay) further add to the end-to-end delay, and are usually only effective on delay variations less than 100 ms. Jitter must therefore be minimized.
  • Latency : Delay for packet delivery Callers usually notice round trip voice delays of 250ms or more. ITU-T G.114 recommends a maximum of a 150 ms one-way latency. Since this includes the entire voice path, part of which may be on the public Internet, your own network should have transit latencies of considerably less than 150 ms.
  • Packet Loss :VOIP is not tolerant of packet loss. Even 1% packet loss can "significantly degrade" a VOIP call using a G.711 codec and other more compressing codecs can tolerate even less packet loss.

    Cisco says:The default G.729 codec requires packet loss far less than 1%
    to avoid audible errors. Ideally, there should be no packet loss for VoIP

- Poor Internet Connection
- Inadequate Router
- Internal Network Improperly Configured

Summary : you must make sure that your wireless connection will never exceed 100 ms jittering and 150 ms one-way latency and 1% packet loss and sure good internet connection and proper network setup

The question here is how we can make sure of that? There are tools in this link which may help in such survey.

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