SDP codec issue during mobile connect

voiphub01 · ‎03-26-2015

Hi Experts,

While using mobility (mobile connect) feature, If i press 'Mobility' key to transfer the call to my mobile, I have observed that CUCM sends G.723 and G.729 codecs in SDP offer. Gateway respective dial-peer is configured to use only G.711Mu codec hence call is rejected by gateway with cause 488 Not Acceptable.

My question is; SIP Trunk, phone which is using mobility and phone who called mobility phone all are using same device pool and configured with 64 Kbps of bandwidth with G.711 as priority 1, then why call manager is sending different codec's in SDP offer.

Thanks in advance.

Regards,

Vip

FYI, following is the SIP trace when mobility key is pressed;

Mar 27 06:18:19.394: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:911@192.168.15.32:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKa2b24e341
From: "Sumer Mehra" <sip:503@192.168.15.11>;tag=310~233834fe-a72b-4338-a652-2dcca84d7d54-38358841
To: <sip:911@192.168.15.32>
Date: Fri, 27 Mar 2015 06:18:19 GMT
Call-ID: e898d00-5141f62b-61-b0fa8c0@192.168.15.11
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
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
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; gci= 0-0
Cisco-Guid: 0243895552-0000065536-0000000075-0185575616
P-Charging-Vector: icid-value="0E898D00000100000000004A0B0FA8C0";icid-generated-at=sub;orig-ioi="IMS Inter Operator Identification"
Session-Expires: 1800
P-Asserted-Identity: "Sumer Mehra" <sip:503@192.168.15.11>
Remote-Party-ID: "Sumer Mehra" <sip:503@192.168.15.11>;party=calling;screen=yes;privacy=off
Contact: <sip:503@192.168.15.11:5060>;isFocus
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 272

v=0
o=CiscoSystemsCCM-SIP 310 1 IN IP4 192.168.15.11
s=SIP Call
c=IN IP4 192.168.15.11
b=TIAS:8000
b=AS:8
t=0 0
m=audio 24612 RTP/AVP 18 4 101
a=rtpmap:4 G723/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Mar 27 06:18:19.410: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKa2b24e341
From: "Sumer Mehra" <sip:503@192.168.15.11>;tag=310~233834fe-a72b-4338-a652-2dcca84d7d54-38358841
To: <sip:911@192.168.15.32>;tag=5A65BB0-5D1
Date: Fri, 27 Mar 2015 06:18:19 GMT
Call-ID: e898d00-5141f62b-61-b0fa8c0@192.168.15.11
CSeq: 101 INVITE
Allow-Events: telephone-event
Warning: 304 192.168.15.32 "Media Type(s) Unavailable"
Reason: Q.850;cause=65
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Mar 27 06:18:19.422: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:911@192.168.15.32:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKa2b24e341
From: "Sumer Mehra" <sip:503@192.168.15.11>;tag=310~233834fe-a72b-4338-a652-2dcca84d7d54-38358841
To: <sip:911@192.168.15.32>;tag=5A65BB0-5D1
Date: Fri, 27 Mar 2015 06:18:19 GMT
Call-ID: e898d00-5141f62b-61-b0fa8c0@192.168.15.11
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

Router#

voiphub01 · ‎03-27-2015

Hi CSC friends,

Further to this, as I have to use only G711 in my 2811 gateway, I changed the settings in CUCM from early offer to delay offer.

This solves the issue to some extent but created other issue of SIP SDP negotiation error in call manager.

When mobility user invoke the moblity key, now call manager is sending INVITE w/o SDP and gateway is responding with SDP offer, however when PSTN user (911) answers the call and gateway sends 200 OK with SDP offer, call manager is not sending SDP answer back in SIP ACK hence call gets disconnected.

Please help Ayodeji, Aaron, Aman, Terry, Chris and my other friends if you are aware about this issue.

Regards,

Vip

SIP traces for your info;

Mar 27 11:32:45.920: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:911@192.168.15.32:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKf263695e75
From: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
To: <sip:911@192.168.15.32>
Date: Fri, 27 Mar 2015 11:32:46 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; gci= 0-0, <sip:192.168.15.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 4230301824-0000065536-0000000129-0185575616
P-Charging-Vector: icid-value="FC25488000010000000000800B0FA8C0";icid-generated-at=sub;orig-ioi="IMS Inter Operator Identification"
Session-Expires: 1800
P-Asserted-Identity: <sip:503@192.168.15.11>
Remote-Party-ID: <sip:503@192.168.15.11>;party=calling;screen=yes;privacy=off
Contact: <sip:503@192.168.15.11:5060>;isFocus
Max-Forwards: 70
Content-Length: 0

Mar 27 11:32:45.948: //123/FC2548800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKf263695e75
From: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
To: <sip:911@192.168.15.32>
Date: Fri, 27 Mar 2015 11:32:45 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

Mar 27 11:32:45.984: //123/FC2548800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKf263695e75
From: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
To: <sip:911@192.168.15.32>;tag=6C64014-BA5
Date: Fri, 27 Mar 2015 11:32:45 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:911@192.168.15.32>;party=called;screen=no;privacy=off
Contact: <sip:911@192.168.15.32:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 9125 1927 IN IP4 192.168.15.32
s=SIP Call
c=IN IP4 192.168.15.32
t=0 0
m=audio 19284 RTP/AVP 0
c=IN IP4 192.168.15.32
a=rtpmap:0 PCMU/8000
a=ptime:20

Mar 27 11:32:46.052: //123/FC2548800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKf263695e75
From: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
To: <sip:911@192.168.15.32>;tag=6C64014-BA5
Date: Fri, 27 Mar 2015 11:32:45 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:911@192.168.15.32>;party=called;screen=no;privacy=off
Contact: <sip:911@192.168.15.32:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 9125 1927 IN IP4 192.168.15.32
s=SIP Call
c=IN IP4 192.168.15.32
t=0 0
m=audio 19284 RTP/AVP 0
c=IN IP4 192.168.15.32
a=rtpmap:0 PCMU/8000
a=ptime:20

Mar 27 11:32:49.080: //123/FC2548800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKf263695e75
From: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
To: <sip:911@192.168.15.32>;tag=6C64014-BA5
Date: Fri, 27 Mar 2015 11:32:45 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:911@192.168.15.32>;party=called;screen=no;privacy=off
Contact: <sip:911@192.168.15.32:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Require: timer
Session-Expires: 1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 194

v=0
o=CiscoSystemsSIP-GW-UserAgent 9125 1927 IN IP4 192.168.15.32
s=SIP Call
c=IN IP4 192.168.15.32
t=0 0
m=audio 19284 RTP/AVP 0
c=IN IP4 192.168.15.32
a=rtpmap:0 PCMU/8000
a=ptime:20

Mar 27 11:32:49.128: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:911@192.168.15.32:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKf3619417d6
From: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
To: <sip:911@192.168.15.32>;tag=6C64014-BA5
Date: Fri, 27 Mar 2015 11:32:46 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

Mar 27 11:32:49.136: //123/FC2548800000/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:503@192.168.15.11:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.32:5060;branch=z9hG4bK1D1DA2
From: <sip:911@192.168.15.32>;tag=6C64014-BA5
To: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
Date: Fri, 27 Mar 2015 11:32:49 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1427455969
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=0
Content-Length: 0

Mar 27 11:32:49.140: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:911@192.168.15.32:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKf4f6937af
From: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
To: <sip:911@192.168.15.32>;tag=6C64014-BA5
Date: Fri, 27 Mar 2015 11:32:46 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
User-Agent: Cisco-CUCM9.1
Max-Forwards: 70
P-Asserted-Identity: <sip:503@192.168.15.11>
CSeq: 102 BYE
Reason: Q.850;cause=47
Content-Length: 0

Mar 27 11:32:49.148: //123/FC2548800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.11:5060;branch=z9hG4bKf4f6937af
From: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
To: <sip:911@192.168.15.32>;tag=6C64014-BA5
Date: Fri, 27 Mar 2015 11:32:49 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 BYE
P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=0
Content-Length: 0

Mar 27 11:32:49.152: //123/FC2548800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.32:5060;branch=z9hG4bK1D1DA2
From: <sip:911@192.168.15.32>;tag=6C64014-BA5
To: <sip:503@192.168.15.11>;tag=489~233834fe-a72b-4338-a652-2dcca84d7d54-38359158
Date: Fri, 27 Mar 2015 11:32:49 GMT
Call-ID: fc254880-51513fde-82-b0fa8c0@192.168.15.11
CSeq: 101 BYE
Content-Length: 0

Jonathan Schulenberg · ‎03-27-2015

We would need to see CCM SDL traces to figure out what happened here. Most likely there is either something in the wrong region or a media resource (eg MTP) getting pulled into the call causing the codec list to get constrained.

voiphub01 · ‎03-27-2015

Hi Jon,

Many thanks for look into the issue.

After reading your comments to take cm traces, I came today to take traces and made test call, things have been changed. Call manager is now sending full codecs in SDP offer however no configuration were changed.

1. My question is which region settings are used when mobility key is pressed. I believe that device pool of remote destination profile (of user who pressed mobility key) and device pool of gateway should be checked. Is it?

Here is SDP offer send by call manager in INVITE now;

v=0
o=CiscoSystemsCCM-SIP 666 1 IN IP4 192.168.15.11
s=SIP Call
c=IN IP4 192.168.15.11
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24682 RTP/AVP 0 8 116 18 124 9 104 105 15 3 4 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:60
a=maxptime:60
a=fmtp:116 mode=30
a=rtpmap:124 iSAC/16000
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=ptime:60
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=ptime:60
a=rtpmap:15 G728/8000
a=ptime:30
a=rtpmap:3 GSM/8000
a=ptime:20
a=rtpmap:4 G723/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2. I also want to know that if you noticed, call manager is sending it's own IP address in SDP which forces to flow the media through call manager. I want media to flow directly between phone and gateway. Can you highlight the reason? Although I have not printed the complete offer/answer above but the negotiated codec is G.711Mu (20msec) still the media is going through call manager. I don't think MTP is required there.

3. My second post/issue was related to SDP error where in case of delay offer (call manager sending INVITE w/o offer) and gateway was responding with offer in 183/200 OK, call manager was not giving SDP answer in ACK, similar to above issue, it started working today (call manager is not giving answer in ACK). No configuration changes were made.

Looking forward to hear from you on point 1 and point 2.

Regards,

Vip

voiphub01 · ‎03-27-2015

BTW, attached is the sdl file taken with above scenario...