No SDP in INVITE on IP2IPGW

Question

Hello,
I am testing a new version of IOS, Version 12.3(11)T2, on a 7204 VXR, for interworking H.323 and SIP. I've enabled the interworking in the IOS with the "allow-connections h323 to sip" command. I'm using a 7960 behind a Call Manager to initiate the H.323 calling. My test call-flow is as follows:
CCM 4.02-->H.323-->7204VXR-->--SIP-->NexTone
When I place a test call from the IP phone, the 7204VXR converts the H.323 call & sends out a SIP INVITE--but, without any SDP information. The missing SDP info that is NOT in the INVITE (endpoint IP address, preferred codec, etc) seems to be present in the upstream H.225 cs: setup messaging coming out of the Call Manager. Packet captures look like the CCM is using fast-start.
For some reason, if I use the "CSIM start xxxxxx" command in the IOS to generate a SIP test call directly from the 7204VXR to my NexTone, the SDP information is included in the INVITE.
Has anyone else had any luck with H.323/SIP interworking on the 72xxVXR? Is the omission of the SDP info in the SIP INVITES a bug, or a case of mis-configuration on my part?
Partial debug CCSIP message output is below.
Thank you,
SK
H.323-to-SIP has no SDP:
ENV1.LAB#term mon
ENV1.LAB#
*Feb 10 13:19:50.546: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.31.20.14:5060;branch=z9hG4bK7A28
From: <sip:[email protected]>;tag=23D59850-F33
To: <sip:[email protected]>
Date: Thu, 10 Feb 2005 13:19:50 GMT
Call-ID: [email protected]
Supported: 100rel,timer
Min-SE: 1800
Cisco-Guid: 2151945792-2627313949-654311424-1076826018
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off
Timestamp: 1108041590
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.31.20.14:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
And now, the SDP info shows up with the "csim start command"
ENV1.LAB#csim start 7035551212
csim: called number = 7035551212, loop count = 1 ping count = 0
*Feb 10 13:21:05.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.31.20.14:5060;branch=z9hG4bK7D1C51
From: <sip:10.31.20.14>;tag=23D6BDFC-1BE7
To: <sip:[email protected]>
Date: Thu, 10 Feb 2005 13:21:05 GMT
Call-ID: [email protected]
Supported: 100rel,timer
Min-SE: 1800
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1108041665
Contact: <sip:10.31.20.14:5060>
Call-Info: <sip:10.31.20.14:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 5535 8577 IN IP4 10.31.20.14
s=SIP Call
c=IN IP4 10.31.20.14
t=0 0
m=audio 31922 RTP/AVP 0 101
c=IN IP4 10.31.20.14
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Answer 1

This featurette (CSCdz58191) allows basic calls to be made between SIP and H.323 based networks, that are connected through the PGW 2200. It supports basic voice calls onlyno services from remote endpoint(s) or PGW 2200s. The following known issues exist:
Passing of DTMF does not work.
T.38 FAX does not work.
Call flows that involve receiving a SIP Re-INVITE do not trigger an H.323 ECS Invocation.
Call flows that involve receiving an H.323 ECS do not trigger the PGW to send a SIP Re-INVITE.
INAP redirection commands are not supported for SIP-H.323 or H.323-SIP calls.
http://www.cisco.com/en/US/products/sw/voicesw/ps1913/prod_release_note09186a008022f692.html#wp770756