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I'm using a SILK codec with Asterisk. (Asterisk is on Version 11.19)

The SILK codec seems to be working correctly, but as time progresses the interval between packet transmissions increases. Because if this, telephone calls are interrupted.

A concrete example would be:

PT=SILK, SSRC=0x9EAADCC0, Seq=0, Time=21760
PT=SILK, SSRC=0x9EAADCC0, Seq=1, Time=22080
320

PT=SILK, SSRC=0x9EAADCC0, Seq=6640, Time=2622080
PT=SILK, SSRC=0x9EAADCC0, Seq=6641, Time=2622720
640

Environment:

iPhone(LinphoneLib) -WiFi -Asterisk(11.19)

I would like to ask if this behavior intended or if not, is there anything I can do about it?

Thank you!

kA tA
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1 Answers1

0

You mentioned:

telephone calls are interrupted.

Can you share the output of the Asterisk CLI command core show channel XXX verbose (where XXX is the channel that exhibits this behavior? Specifically the packet loss and jitter info would be helpful. You can press tab while entering the channel name to auto-complete it.

Also, could you share the silk sections of your codecs.conf file (/etc/asterisk/codecs.conf)? It should look something like:

[silk8]
type=silk
samprate=8000
fec=true
packetloss_percentage=10
maxbitrate=20000
dtx=false

You asked:

is there anything I can do about it?

I would try playing with the silk options in codecs.conf to see if toggling fec/dtx or the packet loss tolerance helps your issue. Don't forget to reload asterisk after you change the config files.

QuickDanger
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