CUBE - SIP Early offer fails when calling from CUCM SIP PHONE

So here's one:

For the ones that are familiar with INE WB vol 1 it's basically module 12 point 2.

So we have a SIP trunk from CUCM to CUBE on CorpHQ router and SIP to ITSP.

Calling from the CUCM SIP phone to PSTN Phone

call works just fine BUT after I enter the command  "voice-class sip early-offer forced" under the dial-peer pointing towards ITSP the behaviour is this: the pstn phone is ringing (clid and cnam and all etc) but once I answer the PSTN phone the call on the CUCM IP phone gets droped. No tone or anything. It just gets droped. And call on PSTN phone remains active - it's talking to the CUBE I suppose.

If i remove the command then it gets connected and it works fine.

So this happens ONLY when calling from the CUCM SIP phone. From CUCM Skinny phone I don't have this problem regardless if it's early offer or not.

Any ideas?

Thanks

Comments

  • It sounds like a media resource issue..  What happens if you check "MTP Required" (and reset trunk) on the SIP trunk forcing fast start from CUCM?  Can you post a copy of 'debug ccsip messages' on the CUBE during the failed call?

  • If I check MTP on trunk and leave default of Required Field = G711 then both calls (from SCCP and SIP) fail from the start. It's normal since I have hardcoded G729br8 in incoming voip dial-peer.

    If I set Required Field = G729 then both calls fail once the call is answered from the PSTN phone.

    And as I said initially with MTP required unchecked only SCCP phone call is working.

    Now I actuallty have an update on the issue. I compared the outputs of debug ccsip messages when initiating the call from SIP phone and when initiating it from SCCP phone.

    I will paste only the output from when I actually answer the call on PSTN phone:

     

    FROM SIP PHONE

    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#!==========ABOUT TO ANSWER
    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#
    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3BD141C
    From: "Jack Shepherd" <sip:[email protected]>;tag=61BDB144-17A0
    To: <sip:[email protected]>;tag=24B3388C-1503
    Date: Tue, 22 Nov 2011 21:05:57 GMT
    Call-ID: [email protected]
    Timestamp: 1321995723
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: "Seattle US Phone" <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 272

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3390 12 IN IP4 177.1.254.250
    s=SIP Call
    c=IN IP4 177.1.254.250
    t=0 0
    m=audio 17302 RTP/AVP 18 100
    c=IN IP4 177.1.254.250
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-16
    a=ptime:20

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3BF12A4
    From: "Jack Shepherd" <sip:[email protected]>;tag=61BDB144-17A0
    To: <sip:[email protected]>;tag=24B3388C-1503
    Date: Tue, 22 Nov 2011 21:02:04 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3BD141C
    From: "Jack Shepherd" <sip:[email protected]>;tag=61BDB144-17A0
    To: <sip:[email protected]>;tag=24B3388C-1503
    Date: Tue, 22 Nov 2011 21:05:57 GMT
    Call-ID: [email protected]
    Timestamp: 1321995723
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: "Seattle US Phone" <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 272

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3390 12 IN IP4 177.1.254.250
    s=SIP Call
    c=IN IP4 177.1.254.250
    t=0 0
    m=audio 17302 RTP/AVP 18 100
    c=IN IP4 177.1.254.250
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-16
    a=ptime:20

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.100.168:5060;branch=z9hG4bK4a9d194efd
    From: "Jack Shepherd" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873234
    To: <sip:[email protected]>;tag=61BDB670-2475
    Date: Tue, 22 Nov 2011 21:02:03 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: kpml, telephone-event
    Remote-Party-ID: "Seattle US Phone" <sip:[email protected]>;party=called;screen=yes;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 292

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 9289 7899 IN IP4 177.1.254.1
    s=SIP Call
    c=IN IP4 177.1.254.1
    t=0 0
    m=audio 16656 RTP/AVP 18 101 19
    c=IN IP4 177.1.254.1
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:19 CN/8000
    a=ptime:20

    CorpHQ(config-dial-peer)#
    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3BF12A4
    From: "Jack Shepherd" <sip:[email protected]>;tag=61BDB144-17A0
    To: <sip:[email protected]>;tag=24B3388C-1503
    Date: Tue, 22 Nov 2011 21:02:04 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Date: Tue, 22 Nov 2011 21:02:03 GMT
    From: "Jack Shepherd" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873234
    Allow-Events: presence, kpml
    Content-Length: 237
    To: <sip:[email protected]>;tag=61BDB670-2475
    Content-Type: application/sdp
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 192.168.100.168:5060;branch=z9hG4bK4aa5f5147c9
    CSeq: 101 ACK
    Max-Forwards: 70

    v=0
    o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.100.168
    s=SIP Call
    c=IN IP4 177.1.11.30
    t=0 0
    m=audio 25422 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    BYE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 177.1.254.1:5060;branch=z9hG4bK3C0767
    From: <sip:[email protected]>;tag=61BDB670-2475
    To: "Jack Shepherd" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873234
    Date: Tue, 22 Nov 2011 21:02:19 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Max-Forwards: 70
    Timestamp: 1321995739
    CSeq: 101 BYE
    Reason: Q.850;cause=65
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:

    CorpHQ(config-dial-peer)#SIP/2.0 200 OK
    Date: Tue, 22 Nov 2011 21:02:19 GMT
    From: <sip:[email protected]>;tag=61BDB670-2475
    Content-Length: 0
    To: "Jack Shepherd" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873234
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 177.1.254.1:5060;branch=z9hG4bK3C0767
    CSeq: 101 BYE


    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#

     

     

    FROM SCCP PHONE

    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#!==========ABOUT TO ANSWER
    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#
    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3C110B1
    From: "Hugo Reyes" <sip:[email protected]>;tag=61BF3348-2556
    To: <sip:[email protected]>;tag=24B4BA9C-1361
    Date: Tue, 22 Nov 2011 21:07:36 GMT
    Call-ID: [email protected]
    Timestamp: 1321995822
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: "Seattle US Phone" <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 273

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 654 3019 IN IP4 177.1.254.250
    s=SIP Call
    c=IN IP4 177.1.254.250
    t=0 0
    m=audio 17952 RTP/AVP 18 100
    c=IN IP4 177.1.254.250
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-16
    a=ptime:20

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3C32353
    From: "Hugo Reyes" <sip:[email protected]>;tag=61BF3348-2556
    To: <sip:[email protected]>;tag=24B4BA9C-1361
    Date: Tue, 22 Nov 2011 21:03:43 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.100.168:5060;branch=z9hG4bK4ae5c4e5699
    From: "Hugo Reyes" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873236
    To: <sip:[email protected]>;tag=61BF37C0-255A
    Date: Tue, 22 Nov 2011 21:03:42 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: kpml, telephone-event
    Remote-Party-ID: "Seattle US Phone" <sip:[email protected]>;party=called;screen=yes;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 292

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3071 3811 IN IP4 177.1.254.1
    s=SIP Call
    c=IN IP4 177.1.254.1
    t=0 0
    m=audio 18554 RTP/AVP 18 101 19
    c=IN IP4 177.1.254.1
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:19 CN/8000
    a=ptime:20

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Date: Tue, 22 Nov 2011 21:03:42 GMT
    From: "Hugo Reyes" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873236
    Allow-Events: presence, kpml
    Content-Length: 227
    To: <sip:[email protected]>;tag=61BF37C0-255A
    Content-Type: application/sdp
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 192.168.100.168:5060;branch=z9hG4bK4af36bbb51c
    CSeq: 101 ACK
    Max-Forwards: 70

    v=0
    o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.100.168
    s=SIP Call
    c=IN IP4 177.1.11.32
    t=0 0
    m=audio 17602 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:0
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3C110B1
    From: "Hugo Reyes" <sip:[email protected]>;tag=61BF3348-2556
    To: <sip:[email protected]>;tag=24B4BA9C-1361
    Date: Tue, 22 Nov 2011 21:07:36 GMT
    Call-ID: [email protected]
    Timestamp: 1321995822
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: "Seattle US Phone" <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 273

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 654 3019 IN IP4 177.1.254.250
    s=SIP Call
    c=IN IP4 177.1.254.250
    t=0 0
    m=audio 17952 RTP/AVP 18 100
    c=IN IP4 177.1.254.250
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-16
    a=ptime:20

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:

    CorpHQ(config-dial-peer)#ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3C32353
    From: "Hugo Reyes" <sip:[email protected]>;tag=61BF3348-2556
    To: <sip:[email protected]>;tag=24B4BA9C-1361
    Date: Tue, 22 Nov 2011 21:03:43 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0


    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#

     

     

     

    What i notice different is this:

    for SIP phone I have

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Date: Tue, 22 Nov 2011 19:52:22 GMT
    From: "Jack Shepherd" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873177
    Allow-Events: presence, kpml
    Content-Length: 237
    To: <sip:[email protected]>;tag=617DECD0-1950
    Content-Type: application/sdp
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 192.168.100.168:5060;branch=z9hG4bK46131d826a7
    CSeq: 101 ACK
    Max-Forwards: 70

    v=0
    o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.100.168
    s=SIP Call
    c=IN IP4 177.1.11.30
    t=0 0
    m=audio 23078 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15

    And then imediately after:

    CorpHQ(config-dial-peer)#BYE sip:[email protected]:5060;transport=tcp SIP/2.0
    Via: SIP/2.0/TCP 177.1.254.1:5060;branch=z9hG4bK37818E9
    From: <sip:[email protected]>;tag=617DECD0-1950
    To: "Jack Shepherd" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873177
    Date: Tue, 22 Nov 2011 19:52:32 GMT
    Call-ID: [email protected]
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Max-Forwards: 70
    Timestamp: 1321991552
    CSeq: 101 BYE
    Reason: Q.850;cause=65
    Content-Length: 0

     

    And for the SCCP phone that actually works I have:

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Date: Tue, 22 Nov 2011 19:59:44 GMT
    From: "Hugo Reyes" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873181
    Allow-Events: presence, kpml
    Content-Length: 227
    To: <sip:[email protected]>;tag=6184A64C-87A
    Content-Type: application/sdp
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 192.168.100.168:5060;branch=z9hG4bK46864c93e69
    CSeq: 101 ACK
    Max-Forwards: 70

    v=0
    o=CiscoSyste
    CorpHQ(config-dial-peer)#msCCM-SIP 2000 1 IN IP4 192.168.100.168
    s=SIP Call
    c=IN IP4 177.1.11.32
    t=0 0
    m=audio 27482 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:0

    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15

  • I think this is actually the key:

    It's like the SIP phone - c=IN IP4 177.1.11.30 - wants codec g729 no annexb. And on all voip dial-peers we have codec g729br8.

     

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Date: Tue, 22 Nov 2011 19:52:22 GMT
    From: "Jack Shepherd" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873177
    Allow-Events: presence, kpml
    Content-Length: 237
    To: <sip:[email protected]>;tag=617DECD0-1950
    Content-Type: application/sdp
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 192.168.100.168:5060;branch=z9hG4bK46131d826a7
    CSeq: 101 ACK
    Max-Forwards: 70

    v=0
    o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.100.168
    s=SIP Call
    c=IN IP4 177.1.11.30
    t=0 0
    m=audio 23078 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15

  • here's another output When early offer is not configured on the dial-peer and the call from SIP phone works.

    we can see the same section from my previous comment here also but this time the call actually works.

     

    CorpHQ(config-dial-peer)#!============ANSWER
    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#
    CorpHQ(config-dial-peer)#
    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3C9215B
    From: "Jack Shepherd" <sip:[email protected]>;tag=61CF2634-4E6
    To: <sip:[email protected]>;tag=24C4AD4C-1FF3
    Date: Tue, 22 Nov 2011 21:25:01 GMT
    Call-ID: [email protected]
    Timestamp: 1321996867
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: "Seattle US Phone" <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 286

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 4829 7656 IN IP4 177.1.254.250
    s=SIP Call
    c=IN IP4 177.1.254.250
    t=0 0
    m=audio 19200 RTP/AVP 18 0 100
    c=IN IP4 177.1.254.250
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:0 PCMU/8000
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-16

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 200 OK
    Via: SIP/2.0/TCP 192.168.100.168:5060;branch=z9hG4bK4b572130045
    From: "Jack Shepherd" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873240
    To: <sip:[email protected]>;tag=61CF2ABC-201B
    Date: Tue, 22 Nov 2011 21:21:07 GMT
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: kpml, telephone-event
    Remote-Party-ID: "Seattle US Phone" <sip:[email protected]>;party=called;screen=yes;privacy=off
    Contact: <sip:[email protected]:5060;transport=tcp>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 292

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 7247 1724 IN IP4 177.1.254.1
    s=SIP Call
    c=IN IP4 177.1.254.1
    t=0 0
    m=audio 16434 RTP/AVP 18 101 19
    c=IN IP4 177.1.254.1
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:19 CN/8000
    a=ptime:20

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected]:5060;transport=tcp SIP/2.0
    Date: Tue, 22 Nov 2011 21:21:07 GMT
    From: "Jack Shepherd" <sip:[email protected]>;tag=66c1c268-ac5b-47c7-97f2-2865657902c3-18873240
    Allow-Events: presence, kpml
    Content-Length: 237
    To: <sip:[email protected]>;tag=61CF2ABC-201B
    Content-Type: application/sdp
    Call-ID: [email protected]
    Via: SIP/2.0/TCP 192.168.100.168:5060;branch=z9hG4bK4b62847e16c
    CSeq: 101 ACK
    Max-Forwards: 70

    v=0
    o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.100.168
    s=SIP Call
    c=IN IP4 177.1.11.30
    t=0 0
    m=audio 24892 RTP/AVP 18 101
    a=rtpmap:18 G729/8000
    a=ptime:20
    a=fmtp:18 annexb=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15


    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3C9215B
    From: "Jack Shepherd" <sip:[email protected]>;tag=61CF2634-4E6
    To: <sip:[email protected]>;tag=24C4AD4C-1FF3
    Date: Tue, 22 Nov 2011 21:25:01 GMT
    Call-ID: [email protected]
    Timestamp: 1321996867
    CSeq: 101 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: "Seattle US Phone" <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060>
    Supported: replaces
    Supported: sdp-anat
    Server: Cisco-SIPGateway/IOS-12.x
    Session-Expires:  1800;refresher=uac
    Require: timer
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 286

    v=0
    o=CiscoSystemsSIP-GW-UserAgent 4829 7656 IN IP4 177.1.254.250
    s=SIP Call
    c=IN IP4 177.1.254.250
    t=0 0
    m=audio 19200 RTP/AVP 18 0 100
    c=IN IP4 177.1.254.250
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:0 PCMU/8000
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-16

    //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK3CBA99
    From: "Jack Shepherd" <sip:[email protected]>;tag=61CF2634-4E6
    To: <sip:[email protected]>;tag=24C4AD4C-1FF3
    Date: Tue, 22 Nov 2011 21:21:08 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 256

    v=0
    o=Cis
    CorpHQ(config-dial-peer)#coSystemsSIP-GW-UserAgent 5496 5377 IN IP4 177.1.254.1
    s=SIP Call
    c=IN IP4 177.1.254.1
    t=0 0
    m=audio 18182 RTP/AVP 18 100
    c=IN IP4 177.1.254.1
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-16

    CorpHQ(config-dial-peer)#

  • Yeah, just now looking at it. You are right, the SIP phone says no annex b, while the PSTN is trying to negotiate g729 annex b.  The call ended with a BYE with cause code 65, which means "Bearer capability not implemented, further pointing to a problem with that codec mismatch.

     

  • I don't know if you caught my last comment. How come we see the same message when no early offer is configured and the call works.

  • ^ I'm still thinking about why it worked with slow start..

    Have you tried this command:


    g729 annexb-all


    To configure Cisco IOS Session Initiation Protocol (SIP) gateway to
    treat the G.729br8 codec as superset of G.729r8 and G.729br8 codecs to
    interoperate with the Cisco Unified Communications Manager, use the g729 annexb-all
    command in voice service SIP configuration mode. To return to the
    default global setting for the gateway, where G.729br8 codec represents
    only the G.729br8 codec, use the no form of this command.

     

  • yes I did, here's the relevant info from the CUBE:

     

    CorpHQ(config-dial-peer)#do sh run | s voice service|dial-peer


    voice service voip
     address-hiding
     allow-connections sip to sip
     redirect ip2ip
     sip
      bind control source-interface Loopback0
      bind media source-interface Loopback0
      header-passing error-passthru
      localhost dns:corphqr1.ine.com
      early-offer forced
      midcall-signaling passthru
      g729 annexb-all


    dial-peer voice 1 pots
     incoming called-number .
     direct-inward-dial


    dial-peer voice 10 pots
     translation-profile outgoing STRIP+DNIS
     preference 1
     destination-pattern +.T
     port 0/0/0:23


    dial-peer voice 101 voip
     destination-pattern 2065011...$
     no voice-class sip early-offer forced
     session protocol sipv2
     session target ipv4:192.168.100.168
     incoming called-number .
     dtmf-relay sip-kpml rtp-nte
     codec g729br8


    dial-peer voice 1001 voip
     destination-pattern +.T
     rtp payload-type nse 99
     rtp payload-type nte 100
     voice-class sip localhost dns:corphqr1.ine.com
     voice-class sip dtmf-relay force rtp-nte
     no voice-class sip early-offer forced
     session protocol sipv2
     session target dns:sip1.att.com
     incoming called-number 206501....$
     dtmf-relay rtp-nte
     codec g729br8


    dial-peer voice 1002 voip
     shutdown
     destination-pattern +.T
     rtp payload-type nse 99
     rtp payload-type nte 100
     voice-class sip localhost dns:corphqr1.ine.com
     voice-class sip dtmf-relay force rtp-nte
     voice-class sip early-offer forced
     session protocol sipv2
     session target dns:sip2.att.com
     incoming called-number 206501....$
     dtmf-relay rtp-nte
     codec g729br8

     

  • I think its a problem with your inbound dial-peer, its set for g729 annex b, but the sip phone cant do that.

    on dial-peer voice 101, remove 'codec g729br8' and put in 'voice-class codec 1' .  If we do that, then the 'g729 annexb-all' command should take action going from g729 and g729b

     

    Also add:

    voice class codec 1
     codec preference 1 g711ulaw
     codec preference 2 g729r8
     codec preference 3 g729br8

     

    ===

    If it still doesnt work, lets add in 'debug voip ccapi inout' debug.

  • i answered but my comment was moderated...

  • so I was saying that using the voice class codec on inbound - still not working.

    using vocie class codec on outbound works but all calls are g711.

    so there's a problem with that SIP phone when early offer and g729 annex b is enforced on dial-ppers.

     

  • Solved it!

    So it says here:

    http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns784/841251_2.pdf

    "Cisco Unified IP phones using SIP as the registration protocol (SIP-line) do not support G.729 with annex B."

    BTW that ^ is a great document for configuring CUBE!

    So I thought I had to use transcoder for this since SIP to ITSP G729br8 was mandatory.

    I checked MTP on SIP trunk in CUCM and prefered originating codec to g729b/g729ab. All calls failed. Thought it was a problem with transcoder and so it was. From previous tasks (module with media resources) I hadn't defined all codecs needed in IOS under transcoder profile. Added the G729abr8 and G729br8 and all calls worked now.

    Scotty, thanks for your input!

  • Awesome!

    On Wed, Nov 23, 2011 at 11:32 AM, inforthewin <[email protected]> wrote:

    Solved it!

    So it says here:

    http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns784/841251_2.pdf

    "Cisco Unified IP phones using SIP as the registration protocol (SIP-line) do not support G.729 with annex B."

    So I thought I had to use transcoder for this since SIP to ITSP G729br8 was mandatory.

    I checked MTP on SIP trunk in CUCM and prefered originating codec to g729b/g729ab. All calls failed. Thought it was a problem with transcoder and so it was. From previous tasks (module with media resources) I hadn't defined all codecs need in IOS under transcoder profile. Added the G729abr8 and G729br8 and all calls worked now.

    Scotty, thanks for your input!




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  • How did you make a Transcoder?

    I have been working on the same issues for several days, and could not force Transcoder to transcode the SIP Phones codec to G729br8.

    Can you please post more details how did you transcode SIP Phones to SIP CUBE?

  • I checked MTP on SIP trunk in CUCM and prefered originating codec to g729b/g729ab.

    Media resource group list of the trunk contains the transcoder which uses dspfarm resources from CorpHQ router. I can post the config of the router if you want but I suggest you check the Media resources module for a thorough explanation of using IOS HW resources - MTP, Transcoder, Conf bridge.

    Sip Phone and Trunk were in the same region and had G711 set. The thing is, on the inbound voice dial-peer I had forced codec g729br8. So basically that's why you need a transcoder. And it's invoked by the trunk.

    Tell me if you have any other questions regarding this.

     

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