Hello,
For a while now I have been getting choppy audio both during recording and playback. Im looking for any ideas or suggestions. I cant for the life of me figure this out. Ill edit the post with more as I think of it.
Heres all the information ive collected:
- Quad Core CPU with 64GB RAM with 1GB ETH (Debian GNU/Linux 9.5 (stretch))
- Calls peak around 100.
- Load average is about 1.5 with ~100 calls and < 1 normally as calls average around ~60.
- I have tried 4 different versions of asterisk now with the same problems
- Current version: Asterisk certified/13.21-cert2
- I have performed hours of RAM checks to see if any errors occured. There were none.
- I have been monitoring the disk (8TB Helium) and have had no errors.
- I have 16GB of Swap assigned but its not being used.
- This issue happens on both VoiP and PRI (using both in my system)
- Ive tried all 3 different timing modules and am currently using res_timing_timerfd.so
- I have dramatically reduced the number of modules loaded (see below for list)
- Using ulaw codec.
# vmstat
procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu-----
r b swpd free buff cache si so bi bo in cs us sy id wa st
2 0 0 590212 1744836 27748772 0 0 213 255 2 0 9 9 81 1 0
Current Timing Module
Module Description Use Count Status Support Level
res_timing_timerfd.so Timerfd Timing Interface 161 Running core
1 modules loaded
Currently loaded modules:
Module Description Use Count Status Support Level
app_chanspy.so Listen to the audio of an active channel 0 Running core
app_confbridge.so Conference Bridge Application 40 Running core
app_controlplayback.so Control Playback Application 0 Running core
app_dial.so Dialing Application 0 Running core
app_directory.so Extension Directory 0 Running core
app_echo.so Simple Echo Application 0 Running core
app_macro.so Extension Macros 0 Running core
app_mixmonitor.so Mixed Audio Monitoring Application 13 Running core
app_page.so Page Multiple Phones 0 Running core
app_playback.so Sound File Playback Application 4 Running core
app_senddtmf.so Send DTMF digits Application 0 Running core
app_stack.so Dialplan subroutines (Gosub, Return, etc 0 Running core
app_transfer.so Transfers a caller to another extension 0 Running core
app_verbose.so Send verbose output 0 Running core
bridge_builtin_features.so Built in bridging features 1 Running core
bridge_simple.so Simple two channel bridging module 0 Running core
bridge_softmix.so Multi-party software based channel mixin 18 Running core
chan_dahdi.so DAHDI Telephony w/PRI 6 Running core
chan_sip.so Session Initiation Protocol (SIP) 51 Running core
codec_gsm.so GSM Coder/Decoder 0 Running core
codec_ulaw.so mu-Law Coder/Decoder 316 Running core
format_gsm.so Raw GSM data 0 Running core
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 Running core
format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 40 Running core
func_callerid.so Party ID related dialplan functions (Cal 0 Running core
func_channel.so Channel information dialplan functions 0 Running core
func_dialplan.so Dialplan Context/Extension/Priority Chec 0 Running core
func_periodic_hook.so Periodic dialplan hooks. 1 Running core
func_volume.so Technology independent volume control 0 Running core
pbx_config.so Text Extension Configuration 0 Running core
pbx_spool.so Outgoing Spool Support 0 Running core
res_agi.so Asterisk Gateway Interface (AGI) 59 Running core
res_ari.so Asterisk RESTful Interface 0 Running core
res_crypto.so Cryptographic Digital Signatures 1 Running core
res_fax.so Generic FAX Applications 1 Running core
res_fax_spandsp.so Spandsp G.711 and T.38 FAX Technologies 0 Running extended
res_monitor.so Call Monitoring Resource 1 Running core
res_musiconhold.so Music On Hold Resource 12 Running core
res_rtp_asterisk.so Asterisk RTP Stack 52 Running core
res_smdi.so Simplified Message Desk Interface (SMDI) 1 Running core
res_sorcery_config.so Sorcery Configuration File Object Wizard 0 Running core
res_speech.so Generic Speech Recognition API 0 Running core
res_stasis.so Stasis application support 0 Running core
res_timing_timerfd.so Timerfd Timing Interface 173 Running core
SIP.CONF
[general]
disallow=all
allow=ulaw
jblog=no
jbenable=yes
jbforce=yes
jbimpl=adaptive
context=inbound
dtmfmode=auto
canreinvite=yes
faxdetect=yes
rtpkeepalive=60
srvlookup=yes
directmedia=no
videosupport=no
defaultexpiry=1800
registerattempts=0
session-timers=refuse
nat=force_rport,comedia
insecure=port,invite
DAHDI.CONF
[channels]
context=inbound
switchtype=national
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=no
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
relaxdtmf=yes
;dtmfmode=auto
;toneduration=300
switchtype=national
signalling=em_w
group=1
channel => 1-24