One part of audio garbled on recordings

[Superspeed]

I have a problem that I can't get solved. I have tried Elastix/Freepbx as well as Asterisk version 1.6/1.8/10
The problem is as follows:

When I use a ECN trunk and I do any recording/monitoring of phone calls one side of the conversation is garbled - it sounds like the person is underwater. This hold true for all our installations regardless of what endpoint devices, routers and PBX used.

Both of the people involved in the phone call can hear each other fine. When I do an internal call I can record both parties or monitor the call without a problem.

I tested with a Telfree trunk and I don't have this problem. Calls record fine. I then told ECN this and they say they can't reproduce the problem on their side when recording any of our calls and basically can't help me/

I'm using G729.

Any suggestions?

[morkhans]

You using monitor or mixmonitor to record?
Is the payload size on g729 set correctly in your SIP trunk (ECN need to specify this)?
Are you using the Digium licensed g729 coded or the unlicensed one?

[Superspeed]

You using monitor or mixmonitor to record?
Is the payload size on g729 set correctly in your SIP trunk (ECN need to specify this)?
Are you using the Digium licensed g729 coded or the unlicensed one?

MixMonitor. I have no idea what the payload should be. I can change one paramater on the ECN CMS next to the codec. It is currently on G729:60 . I can change the part of the colon to 20,30 or 60.

Unlicensed one on the test machine I have now.

[morkhans]

MixMonitor. I have no idea what the payload should be. I can change one paramater on the ECN CMS next to the codec. It is currently on G729:60 . I can change the part of the colon to 20,30 or 60.

Unlicensed one on the test machine I have now.

ECN should tell you what the payload must be. Both sides need to match.
Might be worth dropping $10 on one license to test.

[Superspeed]

ECN should tell you what the payload must be. Both sides need to match.
Might be worth dropping $10 on one license to test.

ECN says it is 60ms. I can change that as well to 20 or 30ms. I'll buy a license later today to test. Where do I change the payload size on the PBX? Any idea?

PS - thank you so much for the help. Been struggling with this one!

[Superspeed]

OMG it worked! You are a ROCKSTAR my friend!! If I can ever help you with any VOIP/Mikrotik or anything else let me know.

I foudn out that my default payload on my PBX is 20ms - ECN was set to 60ms. I changed that to 20ms and its PERFECT. I'm so happy right now... WOW

[Superspeed]

I also discovered now that I can simply set allow=g729:60 in my sip trunk and that also fixes it.

[morkhans]

OMG it worked! You are a ROCKSTAR my friend!! If I can ever help you with any VOIP/Mikrotik or anything else let me know.

I foudn out that my default payload on my PBX is 20ms - ECN was set to 60ms. I changed that to 20ms and its PERFECT. I'm so happy right now... WOW

I thought so

Glad I could help.

I also discovered now that I can simply set allow=g729:60 in my sip trunk and that also fixes it.

As I said both sides need to match. So what is set in your trunk must match what's set on their gateway (as you have discovered).

I'm surprised the out of sync payload only affected your call recordings, normally it messes up the audio for the callers as well.

[Superspeed]

Calls in both directions were perfect but monitoring a call or recording (or any "third" leg a call) was garbled. Amazed that this fixed it!

What is the best payload to use (20ms or 60ms) to ensure the least possible bandwidth is used? I have tried to find the info online but it is very vague.

[morkhans]

Calls in both directions were perfect but monitoring a call or recording (or any "third" leg a call) was garbled. Amazed that this fixed it!

What is the best payload to use (20ms or 60ms) to ensure the least possible bandwidth is used? I have tried to find the info online but it is very vague.

It's the amount of data to send per packet which can be adjusted around the size of the link, which is why I just set it to whatever the VoIP supplier dictates as they normally supply the link as well.

Digium says: "For a low-bandwidth G.729a link, you may want to put a bit more data in each packet." - but does low bandwidth mean small link or low utilized link?

There is a nice complicated discussion here: http://www.voiceie.com/cgi-bin/ultim...c;f=7;t=000068 and here: http://www.cisco.com/en/US/tech/tk65...80094ae2.shtml

[XperiAnce]

I have this EXACT same problem. Asterisk 1.8 with Elastix 2.2.0.

You say I just need to use "allow=g729:60"? It's currently "allow=g729"

[Superspeed]

Yes try g729:60. Alternatively use the CMS to change SIP configuration CODEC to G729:20 (if you are using ECN that is). Asterisk is 20ms by default.

[Superspeed]

It's the amount of data to send per packet which can be adjusted around the size of the link, which is why I just set it to whatever the VoIP supplier dictates as they normally supply the link as well.

Digium says: "For a low-bandwidth G.729a link, you may want to put a bit more data in each packet." - but does low bandwidth mean small link or low utilized link?

There is a nice complicated discussion here: http://www.voiceie.com/cgi-bin/ultim...c;f=7;t=000068 and here: http://www.cisco.com/en/US/tech/tk65...80094ae2.shtml

I'll check it out and get back to you. Thanks again!