Problem with g729 call - cant get around it

[h2009]

Hi there,
Ok im still having the problems with the G729 codec calls. It seems 3CX is dropping the call before RTP are sent/received. I've tested my firewall, its all open, all working fine. Any other codec works fine. My SMB licence is active and has 4 G729 lines. I've been in contact with Mondotalk (SIP) and there end is fine. The way 3CX picks the packets up is causing the problem which i can't fix.
I've been trying for about a week now with no luck. Any help would be greatful.

Here is the log from the server:

21:29:53.588 MediaServerReporting::RTPReceiver [MS105000] C:29.2: No RTP packets were received:remoteAddr=212.248.194.5:39798,extAddr=82.24.13.76:9012,localAddr=82.24.13.76:9012
21:29:53.370 MediaServerReporting::RTPReceiver [MS105000] C:29.1: No RTP packets were received:remoteAddr=192.168.0.198:16458,extAddr=82.24.13.76:9010,localAddr=82.24.13.76:9010
21:29:53.370 Call::Terminate [CM503008]: Call(29): Call is terminated
21:29:53.338 Call::Terminate [CM503008]: Call(29): Call is terminated
21:29:53.213 CallCtrl:nLegConnected [CM503007]: Call(29): Device joined: sip:[email protected]:5060
21:29:53.213 CallCtrl:nLegConnected [CM503007]: Call(29): Device joined: sip:[email protected]:5060
21:29:53.041 Line:rintEndpointInfo [CM505003]: Provider:[Mondotalk] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [CityMoon SIP/1.8.0.004] Transport: [sip:192.168.0.200:5060]
21:29:53.041 CallCtrl:nAnsweredCall [CM503002]: Call(29): Alerting sip:[email protected]:5060
21:29:52.104 CallCtrl:nSelectRouteReq [CM503004]: Call(29): Calling: VoIPline:10@[Dev:sip:[email protected]:5060]
21:29:52.104 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected]]
21:29:52.088 Extension:rintEndpointInfo [CM505001]: Ext.117: Device info: Device Identified: [Man: Linksys;Mod: SPA-921;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA921-5.1.8] Transport: [sip:192.168.0.200:5060]
21:29:52.088 CallCtrl:nIncomingCall [CM503001]: Call(29): Incoming call from Ext.117 to [sip:[email protected]


Here is wireshark capture
codec problem UDP.zip - is a small capture of the problem

I'd be so greatful if someone could solve my problem

Thanks.

 

[h2009]

To be honest i wouldn't know where to start.
If you would make a patch for this i would be extremely grateful - more than you could imagine!
I'd be grateful if you could let me know what you decide to choose to do. That way i wont be waiting around for something.

Thanks for you help on clearing this up, either way.

 

[SY]

Ok i've checked with Mondotalk and they are ONLY using g729a - and not G729b.

Could it be a header communications problem between 3CX and mondotalk.

We are using RFC3555 to specify (describe) codec in SDP content.
Could you please ask provider, what RFC we must use to provide description of RTP payload for G.729 codec to be compatible with Mondotalk?

Thanks

 

[h2009]

Ok i've emailed them just awaiting their reply.

 

[SY]

Ok i've emailed them just awaiting their reply.

We are waiting for answer as well.
Thanks a lot

 

[h2009]

Ok this is the reply i got :

“We use SDP description. Our system will not change the codec description of SDP, it will only change the media address and port.

So when it meets different terminals it uses the standard as decided by the terminals. ”

Does this help at all?

Anything else i can do to help you find the info needed?

 

[SY]

Ok this is the reply i got :

“We use SDP description. Our system will not change the codec description of SDP, it will only change the media address and port.

So when it meets different terminals it uses the standard as decided by the terminals. ”

Does this help at all?

Anything else i can do to help you find the info needed?

Ok. It is the fair description. But it doesn't explore reasons of problem related to access to voice mail provided by Mondotalk.

You can perform following test.
1. Register two extensions on your PBX:
a) SNOM 320 or 360
b) Linksys
2. configure both extension as "Support re-INVITE"=on and "PBX delivers audio"=off
3. configure Linksys to support G729a
4. configure SNOM to support ONLY G729A (all entries on RTP page of account should be set to G729A)
5. Make call from Linksys to SNOM
6. Make call from SNOM to Linksys

Could you please make capture of those calls on PBX host, describe how it works and provide information about firmware version of both phones?

Thanks a lot for your information

 

[h2009]

Ok
Snom 320 - latest firmware, linksys spa921 - latest firmware.
all configured like you wanted, all the calls worked fine, no audio problem, voice clear, calls worked both ways first time around.
The linksys was a remote extension, to my second office.

When calling the linksys to snom the ringing soungs was much faster than normal.
But when calling from/to the linksys to other phones on the which were not setup to for g729a only, the audio seemed to cut out mid call/start of call (on the linksys side, although they could hear me fine).

 

[SY]

Ok
Snom 320 - latest firmware, linksys spa921 - latest firmware.
all configured like you wanted, all the calls worked fine, no audio problem, voice clear, calls worked both ways first time around.
The linksys was a remote extension, to my second office.

When calling the linksys to snom the ringing soungs was much faster than normal.
But when calling from/to the linksys to other phones on the which were not setup to for g729a only, the audio seemed to cut out mid call/start of call (on the linksys side, although they could hear me fine).

Nice picture, realy

What does it mean "latest"?

Thanks

 

[h2009]

linksys firmware : 5.1.8
Snom firmware : 7.1.30

so the most recent. All updated last week.

 

[SY]

linksys firmware : 5.1.8
Snom firmware : 7.1.30

so the most recent. All updated last week.

Can you reach your voicemail on Mondotalk using snom phone?

 

[h2009]

No its the same problem when using pure g729a codec.
But if i leave the default methods of codec, i can hear the voicemail line - no problem, on both linksys and snom. when i've looked at the linksys status page, it says connected using g711u.

If i turn on pure g729 on the handsets line drops out right away.

 

[h2009]

So based on everything you know will you be able to make a patch for mondotalk?

 

[SY]

So based on everything you know will you be able to make a patch for mondotalk?

"alternative" question:
So, based on everything you know will mondotalk be able to make a patch for 3cx?

Thanks

 

[h2009]

Lol - already asked for that, and the short reply was no.

But really how can i get around this, if there is anyway?

 

[archie]

Please also consider asking Linksys and Snom about patch to work with Mondotalk

 

[h2009]

har har - really can it be fixed?

I was looking into reselling mondotalk with 3CX but if g729 calls cant work its not very good.
There are no other voip companies for the UK which are any good (call rates, and line quality)

 

[SY]

Lol - already asked for that, and the short reply was no.

But really how can i get around this, if there is anyway?

I cannot imagine how to support (provide a patch) the system which doesn't plan to support our product.

Thanks for your help and information

 

[h2009]

OK i've actually fixed this problem. Just waiting for mondotalk to call me and confirm it.
Then i'll post how i done it.

 

[zancom]

Hi there,

What have been understanded from 3cx web site that the free 3xc pbx dosen't support the g729.
Can someone confirm that?

Thanks

 

[N_G]

Yes, to use G729 you must use the commercial edition.