ConfBridge audio problems

Asterisk Asterisk Support
1

I’m having major problems using ConfBridge on Asterisk 10.0.0. At seemingly random times during my conferences, the audio will “break up” to the point of being unintelligible. Most of the time, having the speaker disconnect and reconnect to the conference resolves the issue temporarily - but not always. A sample of the bad audio, recorded using MixMonitor, can be heard at http://theblackarrow.net:8880/bad_confbridge_audio.wav

The environment, and a few data points:

  • Asterisk server is at a co-location facility with a public-facing IP address
  • Conferences usually include 4-10 callers, coming in via a mix of SIP softphones and the PSTN (server has a PRI)
  • Softphones used are primarily Jitsi or Acrobits, with either g.722 wideband or g.711 u-law
  • Softphones are generally behind NAT on residential Internet connections
  • Conferences are usually recorded, either with the ConfBridge recorder or MixMonitor on a participant channel
  • Audio problem seems to only happen when a SIP participant is speaking, but isn’t limited to one type of softphone
  • CPU load on the server hovers around 10% during a conference

The SIP channels are configured as follows:

[quote]type=friend
context=
host=dynamic
nat=yes
transport=udp
secret=
callerid=
dtmfmode=rfc2833
disallow=all
allow=g722
allow=ulaw
[/quote]

ConfBridge is configured as follows:

[quote][default_bridge]
type=bridge
max_members=20
internal_sample_rate=16000

[default_user]
type=user
admin=no
marked=no
wait_marked=no
dsp_drop_silence=yes
jitterbuffer=yes
[/quote]

I don’t see any problems indicated on the console at verbosity level 5. Any idea what might be going on here? I haven’t taken the step of actually trying to do SIP or RTP debugging, quite frankly because I wouldn’t know what I’m doing. Thanks in advance for any suggestions.

Josh

2

Are you running Asterisk on VPS or on actual server?

3

It’s an actual server.

4

I have a little bit more information, and a slightly different question.

The audio problems seem to be worst when a particular participant is connected using a softphone (no particular one, he’s tried several) over a particular WiFi connection. If he connects directly to his router using an Ethernet cable, the problem goes away.

Why would this happen? Is there anything that can be tweaked on the router to improve VoIP stability over the WiFi connection?

5

Did you find a fix for this?

I am getting something similar.

At first users join the conference. There is a “Tshhhhh” noise

later into the call it sounds like I am getting some packet loss even when users are on the same LAN segment.

Best regards

6

wifi has high latency, high jitter and may have periods of high packet loss. My guess is that the problem is due to the jitter.

7

Hello,

i have exact the same Problems with confbridge on asterisk 11.12.0 which is running on a virtual maschine. It seems that it happens since we activate G722 within getting some new phones.
Any News about that?

Regards Christian