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SIP to H323 call. Transcoder don't work.

Dear, collegues!

I need your help.

PSTN connected to CME through SIP trunk (early offer). Incoming calls routed to IVR (tcl-service). Most of call from IVR routed to local SCCP phones and some calls routed through GK-controlled H.323 trunk (fast start on both side). I have issue the calls routed through H.323 disconnect after I try off-hook on remote phone (CUCM Phone 701000117) in situation when codec on IVR-dial-peer (711ulaw) and H323(729r8) is different. When I set up on h323-tunk codec similar as on IVR-dial-peer - call established normally. I think that transcoder doesn't work as shall, I don't understand why....Config from CME in attachment.

Somebody can say why doesn't work transcoder between call SIP and H323 call legs when codec is differ?

1 ACCEPTED SOLUTION

Accepted Solutions
Highlighted

Andrei,

I have looked at both traces and I dont see any problem with the transcoder.

The 200 Ok sent to the ITSP has codec g729 in it and the ip address of the CUBE. This suggests to me that the transcoder on the CUBE was invoked...However your ITSP then disconnects the call after Sending the ACK..Which is a little bit weird.

++++++++Here is the 200 Ok sent to the ITSP++++++++++

Sent:    
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.202.0.100:5060;branch=z9hG4bK670622;rport
From: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
To: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
Date: Mon, 17 Dec 2012 14:46:56 GMT
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "Fokin A.V." <117>;party=called;screen=no;privacy=off
Contact: <2220908>
Record-Route: <85.202.0.100:5060>,
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Require: timer
Session-Expires:  3600;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 267

v=0
o=CiscoSystemsSIP-GW-UserAgent 4963 6369 IN IP4 85.202.1.86
s=SIP Call
c=IN IP4 85.202.1.86
t=0 0
m=audio 18934 RTP/AVP 18 101--------------codec used is G729
c=IN IP4 85.202.1.86---------------------------------------->This is the IP address of CUBE (hence media is being terminated here..That suggests xcoding happening
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

+++++++++++Here we recieve ACK from ITSP ( at this point undernormal citcumstances the call should be completed)++++++++++
Received

Dec 17 20:46:59.442 EKT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

ACK sip:2220908@85.202.1.86:5060 SIP/2.0
Via: SIP/2.0/UDP 85.202.0.100:5060;branch=z9hG4bK670626;rport
Max-Forwards: 67
From: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
To: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
Contact:
CSeq: 1 ACK
User-Agent: CommuniGatePro-callLeg/5.4.8
Content-Length: 0

++++++After a few seconds, the call was disconnected...from your ITSP with this reason+++++++++++

Reason: SIP;cause=200;text="bridge closed


Dec 17 20:47:01.010 EKT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:2220908@85.202.1.86:5060;maddr=85.202.1.86 SIP/2.0
Via: SIP/2.0/UDP 85.202.0.100:5060;branch=z9hG4bK670628;rport
Max-Forwards: 67
From: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
To: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
Contact:
CSeq: 3 BYE
Reason: SIP;cause=200;text="bridge closed"
User-Agent: CommuniGatePro-callLeg/5.4.8
Content-Length: 0

In between the call I noticed this....

When CUBE sent an Update to your ITSP notifying them that the called party has changed to extension 117

Dec 17 20:46:56.054 EKT: //66859/6FF4CEE7BFA0/SIP/Msg/ccsipDisplayMsg:
Sent:
UPDATE sip:signode-470054-FAAA460A@85.202.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 85.202.1.86:5060;branch=z9hG4bK3695167C
From: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
To: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
Date: Mon, 17 Dec 2012 14:46:56 GMT
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Route: <85.202.0.100:5060>,
Supported: timer,resource-priority,replaces,sdp-anat
Timestamp: 1355755616
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 UPDATE
Contact: <2220908>
Min-SE:  1800
Remote-Party-ID: "Fokin A.V." <117>;party=called;screen=yes;privacy=off
Content-Length: 0

++++ITSP doesnt seem to like this message..It seems they were expecting something else here.++++++


Dec 17 20:46:56.058 EKT: //66859/6FF4CEE7BFA0/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 85.202.1.86:5060;branch=z9hG4bK3695167C
From: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
To: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
CSeq: 101 UPDATE
Server: CommuniGatePro/5.4.8
Content-Length: 0

My suggestions

Work with your ITSP. Find out why they disconnected the call and what they mean by the disconnect reason

"Reason: SIP;cause=200;text="bridge closed"

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

View solution in original post

4 REPLIES 4
Highlighted
VIP Mentor

Hi Andrei,

The reason you experience this is that Cisco IOS gateway will not invoke a transcoder for any TCL based calls.So you need to use the same codec on the inbound and outbound leg.

Cisco Unified CME B-ACD Limitations

Use same codec on incoming and outgoing dial peers when transferring calls. Using different codecs are not supported. IOS will not invoke transcoder for the calls handled by any TCL application, including B-ACD

From this document...

http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts
Highlighted

Hello, Aokanlawon!

Thank you, your advice may be very useful for me in future.

Yestarday I try turn off my tcl-script and route call from sip call leg to h323 using only translation rule. But I have same result as with tcl.... I attach some debugs (h323 and SIP) for  test call.

Incomig call from bring from 84955803965 (PSTN caller) and translate on CUBE to h323 number 701000117.

Highlighted

Andrei,

I have looked at both traces and I dont see any problem with the transcoder.

The 200 Ok sent to the ITSP has codec g729 in it and the ip address of the CUBE. This suggests to me that the transcoder on the CUBE was invoked...However your ITSP then disconnects the call after Sending the ACK..Which is a little bit weird.

++++++++Here is the 200 Ok sent to the ITSP++++++++++

Sent:    
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.202.0.100:5060;branch=z9hG4bK670622;rport
From: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
To: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
Date: Mon, 17 Dec 2012 14:46:56 GMT
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "Fokin A.V." <117>;party=called;screen=no;privacy=off
Contact: <2220908>
Record-Route: <85.202.0.100:5060>,
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Require: timer
Session-Expires:  3600;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 267

v=0
o=CiscoSystemsSIP-GW-UserAgent 4963 6369 IN IP4 85.202.1.86
s=SIP Call
c=IN IP4 85.202.1.86
t=0 0
m=audio 18934 RTP/AVP 18 101--------------codec used is G729
c=IN IP4 85.202.1.86---------------------------------------->This is the IP address of CUBE (hence media is being terminated here..That suggests xcoding happening
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

+++++++++++Here we recieve ACK from ITSP ( at this point undernormal citcumstances the call should be completed)++++++++++
Received

Dec 17 20:46:59.442 EKT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

ACK sip:2220908@85.202.1.86:5060 SIP/2.0
Via: SIP/2.0/UDP 85.202.0.100:5060;branch=z9hG4bK670626;rport
Max-Forwards: 67
From: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
To: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
Contact:
CSeq: 1 ACK
User-Agent: CommuniGatePro-callLeg/5.4.8
Content-Length: 0

++++++After a few seconds, the call was disconnected...from your ITSP with this reason+++++++++++

Reason: SIP;cause=200;text="bridge closed


Dec 17 20:47:01.010 EKT: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:2220908@85.202.1.86:5060;maddr=85.202.1.86 SIP/2.0
Via: SIP/2.0/UDP 85.202.0.100:5060;branch=z9hG4bK670628;rport
Max-Forwards: 67
From: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
To: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
Contact:
CSeq: 3 BYE
Reason: SIP;cause=200;text="bridge closed"
User-Agent: CommuniGatePro-callLeg/5.4.8
Content-Length: 0

In between the call I noticed this....

When CUBE sent an Update to your ITSP notifying them that the called party has changed to extension 117

Dec 17 20:46:56.054 EKT: //66859/6FF4CEE7BFA0/SIP/Msg/ccsipDisplayMsg:
Sent:
UPDATE sip:signode-470054-FAAA460A@85.202.0.100:5060 SIP/2.0
Via: SIP/2.0/UDP 85.202.1.86:5060;branch=z9hG4bK3695167C
From: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
To: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
Date: Mon, 17 Dec 2012 14:46:56 GMT
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Route: <85.202.0.100:5060>,
Supported: timer,resource-priority,replaces,sdp-anat
Timestamp: 1355755616
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 UPDATE
Contact: <2220908>
Min-SE:  1800
Remote-Party-ID: "Fokin A.V." <117>;party=called;screen=yes;privacy=off
Content-Length: 0

++++ITSP doesnt seem to like this message..It seems they were expecting something else here.++++++


Dec 17 20:46:56.058 EKT: //66859/6FF4CEE7BFA0/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 85.202.1.86:5060;branch=z9hG4bK3695167C
From: <>2220908@sbc.sp-com.ru>;tag=AB0BDE60-2E8
To: "84955803965" <84955803965>;tag=EAFABB17-470054-FAAA460A
Call-ID: 479370ba-c2fb-1230-389f-0583568a5986.gwin
CSeq: 101 UPDATE
Server: CommuniGatePro/5.4.8
Content-Length: 0

My suggestions

Work with your ITSP. Find out why they disconnected the call and what they mean by the disconnect reason

"Reason: SIP;cause=200;text="bridge closed"

Please rate all useful posts

"'Nature is too thin a screen, the glory of the omnipresent God bursts through it everywhere"-Ralph Waldo Emerson

Please rate all useful posts

View solution in original post

Highlighted

Hello, Aokanlawon!

Thank you for help! I try to communicate with service provider.


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