IAX2 trunk problem

Asterisk Asterisk Support
1

HI-
This is crazy - I have two asterisk boxes one is 1.6.2.17.2(BOX A) and the other is 1.6.2.17.20(BOX B) connected by IAX2 trunk

If I reg one of my lines to (BOX B) it sounds great. IF I reg my line to BOX A and go through the IAX trunk to BOX B i hear breaking up of audio(mild) and it doesn’t sound as crisp. I know its not the network layout. when I test it’s after hours and I’m plugged in the same spot [color=#FF0000]closer to Box A[/color]. Box B is located over a VPN connection. But like I said if I reg my line to BOX B (not going through box A ) the audio sounds great. I’m think it’s something to do with the IAX trunk just not sure what to do next.

Thanks!
Joe

2

Some IAX netstat

                       -------- LOCAL ---------------------  -------- REMOTE --------------------

Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts FirstMsg LastMsg
IAX2/MIT-5106 1000 -1 0 -1 -1 0 -1 0 0 0 0 0 0 0 0 Rx:NEW Tx:ACK
1 active IAX channel
localhostCLI> iax2 show netstats
-------- LOCAL --------------------- -------- REMOTE --------------------
Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts FirstMsg LastMsg
IAX2/MIT-5106 1000 -1 0 -1 -1 0 -1 0 0 0 0 0 0 0 0 Rx:NEW Tx:ACK
1 active IAX channel
localhostCLI> iax2 show netstats
-------- LOCAL --------------------- -------- REMOTE --------------------
Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts FirstMsg LastMsg
IAX2/MIT-5106 19 -1 0 -1 -1 0 -1 0 0 40 0 0 0 0 0 Rx:NEW Tx:ACK
1 active IAX channel

3

This is me Reg to Box B (not going through IAX2 trunk) (same location on the network as the IAX trunk call)
filelds don’t line up below

192.168.8.222 1261614288- 00:00:15 0000000758 0000000000 ( 0.00%) 000000 0000000759 0000000000 ( 0.00%) 000001
1 active SIP channel
localhostCLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
192.168.8.222 1261614288- 00:00:16 0000000808 0000000000 ( 0.00%) 000000 0000000807 0000000000 ( 0.00%) 000001
1 active SIP channel
localhostCLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
192.168.8.222 1261614288- 00:00:17 0000000856 0000000000 ( 0.00%) 000000 0000000855 0000000000 ( 0.00%) 000002
1 active SIP channel
localhostCLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
192.168.8.222 1261614288- 00:00:18 0000000904 0000000000 ( 0.00%) 000000 0000000903 0000000000 ( 0.00%) 000001
1 active SIP channel
localhostCLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
192.168.8.222 1261614288- 00:00:19 0000000954 0000000000 ( 0.00%) 000000 0000000954 0000000000 ( 0.00%) 000002
1 active SIP channel
localhostCLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
192.168.8.222 1261614288- 00:00:20 0000000995 0000000000 ( 0.00%) 000000 0000000995 0000000000 ( 0.00%) 000001
1 active SIP channel
localhostCLI> sip show channelstats
Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
192.168.8.222 1261614288- 00:00:20 0000001035 0000000000 ( 0.00%) 000000 0000001035 0000000000 ( 0.00%) 000002
1 active SIP channel

4

Please give us more information: dialplan (extensions.conf), iax.conf, asterisk versions, etc…

5

Hi

firstly when you post output from the cli use the {Code} tags this will preserve the formatting.

now to the problem

You--------BoxA------VPN-----BoxB-----ITSP   (Broken  audio)

You-----------------VPN------BoxB----ITSP    (OK audio)

Is that correct ?

Then you need to look at the codecs that are in use for each leg of the call. as you ma be unnecessarily trans-coding

Ian
www.cyber-cottage.co.uk
twitter @cyberco

6

That’s correct

When you say transcoding - you want me to make sure its ulaw on both ends not ulaw and then gsm on one end?. I just checked everyhting is ulaw(even the iax trunk). I did setup a SIP trunk between the two sites - and everything sounds great on both ends?

Thanks
Joe L

7

I seem to remember either something here, or an issue on the bug tracker, saying that IAX was batching frames.

8

Found this on issue tracker
issues.asterisk.org/view.php?id=18214

9

I’m thinking of something with activity in the last month or two.