Hello Hitesh,

fpiccinin · ‎12-23-2016

Hi all,

we are going to integrate a Cisco Call Manager 9.1.2 vers with a 3rd party SBC.

The requirement are as follow:

- Both audio and video call should be available

- Need to proxy both audio and video on a single IP address

- Need to allow Hold and Resum and Call Transfer

Involved hosts:

172.19.222.84 / 172.19.222.95 CUCM Subscriber

172.19.222.95 Cisco IOS MTP resources

172.25.248.82 SBC

172.25.248.83 SBC Media IP

We have configured a SIP trunk and associated a SIP Profile with the following settings:

SIP Profile:

- Require SDP INactive Exchange... not checked

- RFC 2543 Hold not checked

- Early Offer supporto for voice and video calls (insert MTP if needed) checked

- Send send-receive SDP in mid-call INVITE not checked

ON the MRGL list associated to the trunk we have included only a transcoder type of CISCO IOS Enhanced media Termination Point configured on a 2911 ISR running 15.3.3M5 ios version. It is configured in codec-passthrough.

Transcoder on CUCM is in trusted relay Point mode and this setting is also set as enabled on the trunk.

We have some problem with HOLD and RESUME and it would be fine to have early offer on re-invite for both HOLD and RESUME steps.

Is there a way to set attribute=sendonly and Connect IP = 0.0.0.0 on HOLD reINVITEs?

Do we need to perform SIP Normalization to obtain this? Is there any pre-configured script?

We did some test using both a Jabber or an IP-PHone 7941 running SCCP.41.9-3-1 SR4-1S and we observe this behavior.

CALL SETUP

Call setup from CUCM use early offer (SDP included in the INVITE) but after the ACK on the other party 200OK, CUCM sends another early offer INVITE including only one codec (g711 alaw).

EFFECT: call is up with audio on both party

1st Invite from CUCM:

Internet Protocol Version 4, Src: 172.19.222.85, Dst: 172.25.248.82
User Datagram Protocol, Src Port: 5060, Dst Port: 5080
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:8910@172.25.248.82:5080 SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 172.19.222.85:5060;branch=z9hG4bK65c5c217e419d
        From: <sip:4771@172.19.222.85>;tag=16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243
            SIP from address: sip:4771@172.19.222.85
            SIP from tag: 16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243
        To: <sip:8910@172.25.248.82>
            SIP to address: sip:8910@172.25.248.82
        Date: Fri, 23 Dec 2016 10:37:45 GMT
        Call-ID: d5bb6800-85c1fe79-65b74-55de13ac@172.19.222.85
        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,X-cisco-original-called
        Cisco-Guid: 3585828864-0000065536-0000000225-1440617388
        Session-Expires: 1800
        P-Asserted-Identity: <sip:4771@172.19.222.85>
            SIP PAI Address: sip:4771@172.19.222.85
        Remote-Party-ID: <sip:4771@172.19.222.85>;party=calling;screen=yes;privacy=off
        Contact: <sip:4771@172.19.222.85:5060>
            Contact URI: sip:4771@172.19.222.85:5060
        Max-Forwards: 70
        Content-Type: application/sdp
        Content-Length: 315
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): CiscoSystemsCCM-SIP 16819417 1 IN IP4 172.19.222.85
            Session Name (s): SIP Call
            Connection Information (c): IN IP4 172.19.222.95
            Bandwidth Information (b): TIAS:64000
            Bandwidth Information (b): AS:64
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 18620 RTP/AVP 8 0 18 101
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): ptime:20
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): ptime:20
            Media Attribute (a): rtpmap:18 G729/8000
            Media Attribute (a): ptime:20
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-15

2nd Invite from CUCM

Internet Protocol Version 4, Src: 172.19.222.85, Dst: 172.25.248.82
User Datagram Protocol, Src Port: 5060, Dst Port: 5080
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:8910@172.25.248.83:5080;transport=udp SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 172.19.222.85:5060;branch=z9hG4bK65c5e4f6dd9b0
        From: <sip:4771@172.19.222.85>;tag=16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243
            SIP from address: sip:4771@172.19.222.85
            SIP from tag: 16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243
        To: <sip:8910@172.25.248.82>;tag=D2D2yaHZ40Z4H
            SIP to address: sip:8910@172.25.248.82
            SIP to tag: D2D2yaHZ40Z4H
        Date: Fri, 23 Dec 2016 10:37:46 GMT
        Call-ID: d5bb6800-85c1fe79-65b74-55de13ac@172.19.222.85
        Route: <sip:172.25.248.82:5080;lr;ftag=16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243;did=cf7.f88ab87>
        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: 102 INVITE
        Max-Forwards: 70
        Expires: 180
        Allow-Events: presence
        Supported: X-cisco-srtp-fallback
        Supported: Geolocation
        Session-Expires: 1800;refresher=uac
        P-Asserted-Identity: <sip:4771@172.19.222.85>
            SIP PAI Address: sip:4771@172.19.222.85
        Remote-Party-ID: <sip:4771@172.19.222.85>;party=calling;screen=yes;privacy=off
        Contact: <sip:4771@172.19.222.85:5060>
            Contact URI: sip:4771@172.19.222.85:5060
        Content-Type: application/sdp
        Content-Length: 241
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): CiscoSystemsCCM-SIP 16819417 2 IN IP4 172.19.222.85
            Session Name (s): SIP Call
            Connection Information (c): IN IP4 172.19.222.95
            Bandwidth Information (b): TIAS:64000
            Bandwidth Information (b): AS:64
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 18620 RTP/AVP 8 101
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): ptime:20
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-15

CALL HOLD FROM CISCO SIDE

When we make and Hold from CISCO Side, CUCM send 1st an Early Offer with media inactive attribute and then a Delayed offer including send-only attribute on the ACK to the SBC's 200OK answer.

EFFECT: the holded party DOES NOT HEAR ANY MOH stream.

CUCM HOLD's 1st Early OFFER INVITE

Internet Protocol Version 4, Src: 172.19.222.85, Dst: 172.25.248.82
User Datagram Protocol, Src Port: 5060, Dst Port: 5080
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:8910@172.25.248.83:5080;transport=udp SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 172.19.222.85:5060;branch=z9hG4bK65c646a694d0
        From: <sip:4771@172.19.222.85>;tag=16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243
            SIP from address: sip:4771@172.19.222.85
            SIP from tag: 16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243
        To: <sip:8910@172.25.248.82>;tag=D2D2yaHZ40Z4H
            SIP to address: sip:8910@172.25.248.82
            SIP to tag: D2D2yaHZ40Z4H
        Date: Fri, 23 Dec 2016 10:38:01 GMT
        Call-ID: d5bb6800-85c1fe79-65b74-55de13ac@172.19.222.85
        Route: <sip:172.25.248.82:5080;lr;ftag=16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243;did=cf7.f88ab87>
        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: 103 INVITE
        Max-Forwards: 70
        Expires: 180
        Allow-Events: presence
        Supported: X-cisco-srtp-fallback
        Supported: Geolocation
        Session-Expires: 1800;refresher=uac
        P-Asserted-Identity: <sip:4771@172.19.222.85>
            SIP PAI Address: sip:4771@172.19.222.85
        Remote-Party-ID: <sip:4771@172.19.222.85>;party=calling;screen=yes;privacy=off
        Contact: <sip:4771@172.19.222.85:5060>
            Contact URI: sip:4771@172.19.222.85:5060
        Content-Type: application/sdp
        Content-Length: 247
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): CiscoSystemsCCM-SIP 16819417 3 IN IP4 172.19.222.85
            Session Name (s): SIP Call
            Connection Information (c): IN IP4 0.0.0.0
            Bandwidth Information (b): TIAS:64000
            Bandwidth Information (b): AS:64
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 18620 RTP/AVP 8 101
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): ptime:20
            Media Attribute (a): inactive
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-15

CUCM's ACK to SBC's 200OK on delayed offer INVITE:

Internet Protocol Version 4, Src: 172.19.222.85, Dst: 172.25.248.82
User Datagram Protocol, Src Port: 5060, Dst Port: 5080
Session Initiation Protocol (ACK)
    Request-Line: ACK sip:8910@172.25.248.83:5080;transport=udp SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 172.19.222.85:5060;branch=z9hG4bK65c677ae3fdf
        From: <sip:4771@172.19.222.85>;tag=16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243
            SIP from address: sip:4771@172.19.222.85
            SIP from tag: 16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243
        To: <sip:8910@172.25.248.82>;tag=D2D2yaHZ40Z4H
            SIP to address: sip:8910@172.25.248.82
            SIP to tag: D2D2yaHZ40Z4H
        Date: Fri, 23 Dec 2016 10:38:01 GMT
        Call-ID: d5bb6800-85c1fe79-65b74-55de13ac@172.19.222.85
        Route: <sip:172.25.248.82:5080;lr;ftag=16819417~305cfde1-0d58-4b14-a9ef-7b3502d970ea-39612243;did=cf7.f88ab87>
        Max-Forwards: 70
        CSeq: 104 ACK
        Allow-Events: presence
        Content-Type: application/sdp
        Content-Length: 195
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): CiscoSystemsCCM-SIP 16819417 4 IN IP4 172.19.222.85
            Session Name (s): SIP Call
            Connection Information (c): IN IP4 172.19.222.95
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 4000 RTP/AVP 8
            Media Attribute (a): X-cisco-media:umoh
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): ptime:20
            Media Attribute (a): sendonly

CALL RESUME FROM CISCO SIDE

For the RESUME from CUCM side, it performs the same procedure used for HOLD except the fact that in the delayed offer ack it does not put any attribute in order to resum the media.

EFFECT: no audio at all from/to both parties.

Thanks for Help

Regards

Francesco

hsood · ‎12-25-2016

Hi Francesco,

From the snippets, after the hold, it breaks the already setup media with 0.0.0.0 and then a DO re-invite which in turn has ACK with SDP and I can see the CUCM sending attribute as sendonly, connection information as 172.19.222.95 (CUCM as MOH server ) and port as 4000 which is perfect for streaming MOH to SBC.

Do you have the MOH in the MRGL of the SIP Trunk?

Can you hardcode the SIP trunk with "MTP required" checkbox checked and then test.

Also, if that does not work, please post a packet capture from MOH and CCM trace (set to detailed) from all nodes for a test call on hold.

Regards,

Hitesh

fpiccinin · ‎01-02-2017

Hello Hitesh,

thanks for you quick answer and sorry for my late reply.

By the way I have some question for you:

1. On the other side (SBC cube is connecting to) prefer early offer on re-invite, is it possibile to force it from cube or also from my cucm side?

2. I'm going to add MOH server to MRGL list since it is missing now. Should I expect moh stream flowing from CUCM to SBC or should cube proxy it?

3. If I'm going to hardcode MTP Required on CUCM SIP trunk side, will video calls be supported?

4. If I'm using media flow through on cube, will video calls be supported?

I'm going to test moh and keep you updated.

Thanks

Regards

CISCO SIP TRUNK Early Offer INVITE - ReINVITE