[asterisk-users] 1 second Audio Lag

[colin mcdermott]

Hi All (reposting after 24 hours).

I will do a test call from a soft phone to my mobile. I can speak into my
headset and the audio is heard instantly. But if I speak into my mobile
there is a 1-2 second delay in the Audio. I am using SIP.

I am only finding it in the Zoiper Softphones that we are using. We are able
to make a call without lag on the X-lite softphone no problem. Sadly the
customer is Quite attached to the Zoiper.

I have set QOS = CS5 for both SIP and RTP packets. Altering these settings
has no effect to the lag issue.

We have three 24 port Gigabit switches, with the top switch connecting in
the Asterisk Box. Even the stations plugged into the TOP switch have this
delay and to the same extent as the other switches. No routers on the loop

I have tried switching the stations to IAX. No effect. I have tried using
GSM instead of G711 (alaw). No effect. I have about 30 stations. No change
under heavy or light load.

I have done a Wireshark trace on the stations and no issues detected when I
go analyse on the RTP packets. All sequencing is correct.

Is Zoiper any good? Anyone else had these problems?
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