We’ve implemented a Stasis app with ARI that in one of the use cases it sends messages to a destination by making playbacks on that channel.
In a virtual machine with 1 CPU (it’s not very different in a 4 CPUs configuration) it can’t do more than 5 simultaneous calls without pulling the CPU to 100%, obviuously causing degradation on the audio quality.
The peer used to make the calls is configured to use only the “a-law” codec (disallow=all allow=alaw), and the audio file is encoded with “a-law”. In the channels list I can see that the channel is using “a-law”, for the SIP channel:
host*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
IP1 3512ABCD 429796c350f9c14 (alaw) No Tx: ACK Trunk-T1
But there are 3 channels:
host*CLI> core show channels
Channel Location State Application(Data)
Local/100@CTX-0000 100@CTXT:4 Up Dial(SIP/Trunk-T1/3512ABCD
Local/100@CTX-0000 100@CTXT:1 Up Stasis(app_name)
SIP/Trunk-TB1-00 (None) Up AppDial((Outgoing Line))
3 active channels
1 active call
This is because I originate the call in ARI to “Local/100@CTXT”, and in context CTXT there is this dial:
same => n,Dial(SIP/Trunk-T1/${callednum},${ctimeout})
The most obvious reason for this behaviour would be that asterisk would be making transcoding, which is the case, as we can see in the transcoding information on this channels:
For Local channel 1:
host*CLI> core show channel Local/100@CTXT-00000000;1
– General –
Type: Local
(…)
NativeFormats: (slin)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: Yes (slin@8000)->(slin@192000)
ReadTranscode: No
For Local channel 2:
hostCLI> core show channel Local/100@G9_TB-00000000;2
– General –
(…)
NativeFormats: (slin)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: No
ReadTranscode: No
For the SIP channel:
host*CLI> core show channel SIP/Trunk-T1-00000000
– General –
Type: SIP
NativeFormats: (alaw)
WriteFormat: slin192
ReadFormat: slin192
WriteTranscode: Yes (slin@192000)->(slin@8000)->(alaw@8000)
ReadTranscode: Yes (alaw@8000)->(slin@8000)->(slin@192000)
Is it possible to configure the Local channel to use alaw, as I did in the trunk T1?
As I also need to create bridges between the channels, I can’t originate a channel to SIP, as the bridges don’t work on this case.
Asterisk version is 13.19.2 (upgraded from 13.4.0), running on CentOS 6.