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samer.sabeeh
Beginner

CUCM 8.5 SIP initial invite SDP contains "a=sendonly"

Hello,

we have cucm 6.1 and we are about to upgrade to 8.5 so i have setup a lab enviroment,

all work as expected except outgoing sip calls to IP2IP GW,CUCM send a=sendonly in an INVITE messages causing one way sort of call.

reseting the trunk seams to fix the issue for temporary period and it start to happen all again,

SDI trace is attached for more details, kindly enlighten me with your opinions

regards

1 ACCEPTED SOLUTION

Accepted Solutions

I think we are hitting http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCtk77040

Symptom:
MTP resources may be intermittently leaked, ultimately resulting in failure of
SIP calls that require MTP resources.  From RTMT, available MTP resources reach
0 and MTP allocation failure counts go up for each call requiring an MTP.
The SDP portion of the initial INVITE will incorrectly contain "a=inactive".

Conditions:
This issue can only happen in an outgoing initial SIP call setup where MTP is
required.  In this case, the outgoing SIP INVITE message will contain an SDP
offer.  The issue may occur in the following scenarios:
- Outgoing SIP trunk calls with "Media Termination Point Required" checked on
the SIP trunk.
- Calls between IPv6-only endpoints and IPv4-only endpoints.

This problem does NOT occur in the following scenarios:
- outgoing delayed-offer SIP trunk calls, even if an MTP is required.
- incoming SIP trunk calls.
- mid-call incoming or outgoing INVITE messages.

Workaround:
1. Uncheck "Media Termination Point Required" on the SIP Trunk configuration if
possible.
2. If Early Offer is required, configure Early Offer but leave "Media
Termination Point Required" unchecked.
3. For IPv6 deployment, use dual-stack rather than IPv6-only endpoints.


5229,NET]
INVITE sip:0633703817@192.168.10.254:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.10.10:5060;branch=z9hG4bK2394f67ba
From: "Test" <101>;tag=2572~43746739-4302-4c01-ade9-75ccfae8f6eb-18033884
To: <0633703817>
Date: Mon, 21 Mar 2011 09:38:47 GMT
Call-ID: 3c97300-d8711ca7-e-a0aa8c0@192.168.10.10
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0063533824-0000065536-0000000014-0168470720
Session-Expires:  1800
P-Asserted-Identity: "Test" <101>
Remote-Party-ID: "Test" <101>;party=calling;screen=yes;privacy=off
Contact: <101>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 227

v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.10.10
s=SIP Call
c=IN IP4 192.168.10.254
t=0 0
m=audio 17320 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly ===> This is the problem :-)
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

View solution in original post

3 REPLIES 3
samer.sabeeh
Beginner

any one care to help ?

I think we are hitting http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCtk77040

Symptom:
MTP resources may be intermittently leaked, ultimately resulting in failure of
SIP calls that require MTP resources.  From RTMT, available MTP resources reach
0 and MTP allocation failure counts go up for each call requiring an MTP.
The SDP portion of the initial INVITE will incorrectly contain "a=inactive".

Conditions:
This issue can only happen in an outgoing initial SIP call setup where MTP is
required.  In this case, the outgoing SIP INVITE message will contain an SDP
offer.  The issue may occur in the following scenarios:
- Outgoing SIP trunk calls with "Media Termination Point Required" checked on
the SIP trunk.
- Calls between IPv6-only endpoints and IPv4-only endpoints.

This problem does NOT occur in the following scenarios:
- outgoing delayed-offer SIP trunk calls, even if an MTP is required.
- incoming SIP trunk calls.
- mid-call incoming or outgoing INVITE messages.

Workaround:
1. Uncheck "Media Termination Point Required" on the SIP Trunk configuration if
possible.
2. If Early Offer is required, configure Early Offer but leave "Media
Termination Point Required" unchecked.
3. For IPv6 deployment, use dual-stack rather than IPv6-only endpoints.


5229,NET]
INVITE sip:0633703817@192.168.10.254:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.10.10:5060;branch=z9hG4bK2394f67ba
From: "Test" <101>;tag=2572~43746739-4302-4c01-ade9-75ccfae8f6eb-18033884
To: <0633703817>
Date: Mon, 21 Mar 2011 09:38:47 GMT
Call-ID: 3c97300-d8711ca7-e-a0aa8c0@192.168.10.10
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0063533824-0000065536-0000000014-0168470720
Session-Expires:  1800
P-Asserted-Identity: "Test" <101>
Remote-Party-ID: "Test" <101>;party=calling;screen=yes;privacy=off
Contact: <101>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 227

v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.10.10
s=SIP Call
c=IN IP4 192.168.10.254
t=0 0
m=audio 17320 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly ===> This is the problem :-)
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

View solution in original post

thanks for the reply,

so if MTP resources are available, this is most likely a BUG

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