Jitter Buffer / Sound Break / Dealy in Voice [RESOLVED]

Asterisk Asterisk Support
1

Hi.
I’m currently using asterisk to communicate with remote office asterisk box. The latency between both server remains between 150 - 350ms. (sometimes it get worse). I’m using jitter buffer to cut off the delay in voice / sound break.

The configuration i’m using for jitter buffer in iax.conf is

tos=ef jitterbuffer=yes maxjitterbuffer=300 maxjitterinterps=10 resyncthreshold=1400 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1

but still having issues with sound below is the netstats of iax route

-------- LOCAL ---------------------  -------- REMOTE --------------------
Channel               RTT  Jit  Del  Lost   %  Drop  OOO  Kpkts  Jit  Del  Lost   %  Drop  OOO  Kpkts FirstMsg    LastMsg
IAX2/UKStaff2-17259 1000   75  134     4   0     3    4      0    0    0     0   0     0    0      0 Tx:NEW      Rx:ACK
IAX2/UKStaff2-22059  221   71  123    26   2     8   59      0    0   40     0   0     0    0      0 Tx:NEW      Rx:ACK
IAX2/UKStaff2-23122 1000   85  130    13   2     6   22      0    0    0     0   0     0    0      0 Tx:NEW      Tx:LAGRQ
IAX2/UKStaff2-23699 1000   75  122    22   2    11   70      0    0    0     0   0     0    0      0 Tx:NEW      Tx:PING
IAX2/UKStaff2-24499 1000   75  135     4   0     2    3      0    0    0     0   0     0    0      0 Tx:NEW      Rx:ACK
IAX2/UKStaff2-27974 1000   75  122     6   1     3   24      0    0    0     0   0     0    0      0 Tx:NEW      Tx:ACK
IAX2/UKStaff2-28518 1000   75  116     6   1     3   41      0    0    0     0   0     0    0      0 Tx:NEW      Tx:HANGUP

Currently all channels are using g729 for communication any suggestion i can improve the sound quality ?

Looking forward to your kind response.
Thank you.

2

Prioritise IAX traffic across the whole network and/or provide a network with enough capacity to support the offered traffic.

3

Dear David,
Thank you for your response. I have tried it via TC but nothing seems working do you any reference script i can try ?

BTW the asterisk is hosted on a dedicated server & the other is in remote office.

4

You will need to talk to the people who operate the network you are using, as differentiated service markings are unlikely to be honoured once you leave the part of the network that you manage. I’m assuming the connection is IP, rather than a leased line.

Your only option may be to use an alternative supplier. If you are in a country with a monopoly supplier, you may have to use the PSTN, as the only network specifically designed for voice traffic.

5

Note: I’m looking at your lost packet counts, rather than the jitter.

6

Acknowledged for testing do you have a reference for QOS script ?

7

svn.digium.com/svn/asterisk/bran … onf.sample

8

You need to read this digium.com/en/solutions/ip-p … of-service