[asterisk-users] SIP jitter and packlost channel variables

[Arjan Kroon | Mobillion]

Hi,

A client of ours get lots of problem with there voice quality when the do a lot SIP calls.
In a application I log the rtpqos audio jitter an lost packets.  (see Below)

Does anybody know what the numbers mean?
If I look at a sample of the channel variables, I see the following number.
local_lostpackets = 7706
local_jitter = 2
local_maxjitter = 11
local_minjitter = 0
..
..
remote_lostpackets = 0
remote_jitter = 0
remote_maxjitter = 70000
remote_minjitter = 14000
..
..

The only thing I see is this: http://www.voip-info.org/wiki/view/Asterisk+func+channel

Regards,

Arjan Kroon
Mobillion BV


exten => s,n,Set(A_SIP_DATA=${CHANNEL(rtpqos,audio,local_lostpackets)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_jitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_maxjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_minjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_normdevjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,local_stdevjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_lostpackets)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_jitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_maxjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_minjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_normdevjitter)})
exten => s,n,Set(A_SIP_DATA=${A_SIP_DATA},${CHANNEL(rtpqos,audio,remote_stdevjitter)})