troubleshooting sip with cisco unified communications · • session initiation protocol (sip)...
TRANSCRIPT
BRKUCC-2932
Troubleshooting SIP with Cisco Unified
Communications
Paul Giralt (@PaulGiralt)
Distinguished Services Engineer
Agenda
• Introduction
• Session Initiation Protocol (SIP) Overview
• Troubleshooting Tools
• Unified CM Tracing
• Cisco Unified Border Element (CUBE) Tracing
• Sample Call Flows / Case Studies
What is SIP?
• Signaling protocol used to establish, manage, and terminate sessions over an IP network
• Core protocol defined in RFC 3261
• Extended in many, many other RFCs
• ASCII-based messages
• Endpoints are referred to as User Agents
What is SIP?
• User Agents
• SIP Messages
• Requests and Responses
• Headers
• Media Negotiation
• Session Description Protocol
• Offer/Answer Model
• Early Offer vs. Delayed Offer
• Early Media
• DTMF Relay
User Agents
• User Agent Clients (UAC) send requests to User Agent Servers (UAS)
• User Agent Servers send responses to the requests
• Most SIP devices are both a UAC and a UAS (they both initiate and accept requests)
• Unified CM and CUBE are both Back-to-Back User Agents (B2BUA) (as opposed to Proxies)
• Cisco VCS and Cisco Expressway operate as both proxies and B2BUA’s
SIP Request Methods from RFC 3261
• INVITE - A user or service is being invited to participate in a multimedia session
• ACK - Confirms that a client has received a final response to an INVITE request
• BYE - Terminates an existing session; can be sent by any user agent (in a multiparty session)
• CANCEL - Cancels pending requests; does not terminate sessions that have been accepted
• OPTIONS - Queries the capabilities of servers (Also used as a keep alive)
• REGISTER - Registers the user agent with the registrar server of a domain
Additional SIP Request Methods
• INFO (RFC 2976) - to send more information within an established dialog
• PRACK (RFC 3262) - to acknowledge a provisional response
• SUBSCRIBE (RFC 3265) - to tell a remote node to look for a certain event
• NOTIFY (RFC 3265) - to respond when that certain event occurs
• UPDATE (RFC 3311) - to update parameters of a session set-up
• MESSAGE (RFC 3428) - SIP instant messaging
• REFER (RFC 3515) – to “refer” one UA to communicate with another UA
• PUBLISH (RFC 3903) - to push UA state information to a compositor/presence server
SIP INVITE MethodINVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665From: "Test User 1" <sip:[email protected]>;tag=97903bc0-43adcd-45510543To: <sip:[email protected]>Call-ID: [email protected]: timer,resource-priority,replacesUser-Agent: Cisco-CUCM8.6Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFERCSeq: 101 INVITEExpires: 180Allow-Events: presence, kpmlSupported: X-cisco-srtp-fallbackSupported: GeolocationCall-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"Cisco-Guid: 2081204224-3137452793-0000000466-0996807340Session-Expires: 1800P-Asserted-Identity: "Test User 1" <sip:[email protected]>Contact: <sip:[email protected]:5060>;video;audioMax-Forwards: 69Content-Length: 864Content-Type: application/sdp
SIP Request LineINVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665From: "Test User 1" <sip:[email protected]>;tag=97903bc0-43adcd-45510543To: <sip:[email protected]>Call-ID: [email protected]: timer,resource-priority,replacesUser-Agent: Cisco-CUCM8.6Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFERCSeq: 101 INVITEExpires: 180Allow-Events: presence, kpmlSupported: X-cisco-srtp-fallbackSupported: GeolocationCall-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"Cisco-Guid: 2081204224-3137452793-0000000466-0996807340Session-Expires: 1800P-Asserted-Identity: "Test User 1" <sip:[email protected]>Contact: <sip:[email protected]:5060>;video;audioMax-Forwards: 69Content-Length: 864Content-Type: application/sdp
SIP Method
URI SIP Version
SIP HeadersINVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665From: "Test User 1" <sip:[email protected]>;tag=97903bc0-43adcd-45510543To: <sip:[email protected]>Call-ID: [email protected]: timer,resource-priority,replacesUser-Agent: Cisco-CUCM8.6Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFERCSeq: 101 INVITEExpires: 180Allow-Events: presence, kpmlSupported: X-cisco-srtp-fallbackSupported: GeolocationCall-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"Cisco-Guid: 2081204224-3137452793-0000000466-0996807340Session-Expires: 1800P-Asserted-Identity: "Test User 1" <sip:[email protected]>Contact: <sip:[email protected]:5060>;video;audioMax-Forwards: 69Content-Length: 864Content-Type: application/sdp
SIP ResponseSIP/2.0 404 Not FoundVia: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4From: "Paul Giralt" <sip:[email protected]>;tag=19210123ca7-45568313To: <sip:[email protected]>;tag=253488-726Date: Mon, 16 Jan 2012 04:00:22 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.2.TReason: Q.850;cause=1Content-Length: 0
SIP ResponseSIP/2.0 404 Not FoundVia: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4From: "Paul Giralt" <sip:[email protected]>;tag=19210123ca7-45568313To: <sip:[email protected]>;tag=253488-726Date: Mon, 16 Jan 2012 04:00:22 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.2.TReason: Q.850;cause=1Content-Length: 0
Response Code
Free-text Reason
SIP ResponseSIP/2.0 404 Not FoundVia: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4From: "Paul Giralt" <sip:[email protected]>;tag=19210123ca7-45568313To: <sip:[email protected]>;tag=253488-726Date: Mon, 16 Jan 2012 04:00:22 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: telephone-eventServer: Cisco-SIPGateway/IOS-15.2.2.TReason: Q.850;cause=1Content-Length: 0
SIP ResponsesResponse Code Description Example
1xx Informational – Request Received and Continuing to Process
Request
100 Trying
180 Ringing
183 Session Progress
2xx Success – Action was successfully received, understood, and
accepted
200 OK
202 Acceptable
3xx Redirection – Another SIP Element needs to be contacted in order
to complete the request
300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
4xx Client Error – Request contains bad syntax or cannot be fulfilled at
this server
401 Unauthorized
404 Not Found
406 Not Acceptable
486 Busy Here
488 Not Acceptable Here
5xx Server Error – Server failed to fulfill an apparently valid request 503 Service Unavailable
6xx Global Failure – Request is invalid at any server 600 Busy Everywhere
603 Decline
INVITE
200 OK
Session Established
Phone 1 Unified CM
Basic SIP Call Setup with B2BUA (Unified CM)
ACK
BYE
200 OK
Phone 2
INVITE
200 OK
ACK
BYE
200 OK
CUBE
SBC (CUBE)
INVITE
200 OK
Phone 1
Basic SIP Call Setup with Unified CM and CUBE
ACK
BYE
200 OK
Unified CM
CUBE
SBC (CUBE)
INVITE
200 OK
ACK
BYE
200 OK
INVITE
200 OK
ACK
BYE
200 OK
SIP
SPSBC
SP SBC
Session Established
Media Negotiation
• SIP leverages the Session Description Protocol (SDP) (RFC 4566/3266/2327) to communicate media information.
• SIP uses the offer/answer model described in RFC 3264 to negotiate media using SDP
Offer/Answer Model (RFC 3264)
• One endpoint sends an offer SDP containing all the capabilities the endpoint wishes to negotiate.
• SDP contains m lines for each media stream being negotiated (i.e. audio, video, content channel, etc…)
• Receiving endpoint sends an answer SDP that contains the same or a subset of capabilities received in the offer.
• Per RFC 3264, “For each "m=" line in the offer, there MUST be a corresponding "m=“ line in the answer. The answer MUST contain exactly the same number of "m=" lines as the offer.”
Session Description Protocol (SDP) - Offerv=0
o=Cisco-SIPUA 26964 0 IN IP4 172.18.159.152
s=SIP Call
t=0 0
m=audio 29254 RTP/SAVP 0 8 18 102 9 116 124 101
c=IN IP4 172.18.159.152
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 25466 RTP/AVP 97
c=IN IP4 172.18.159.152
b=TIAS:1000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801E
a=recvonly
Session Description Protocol (SDP) - Answer
v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.18.106.59
s=SIP Call
c=IN IP4 172.18.159.152
t=0 0
m=audio 30308 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 97
Media Negotiation – Early Offer and Delayed Offer
• Initiator of the call can send SDP offer in the INVITE – this is called an Early Offer (EO)
• Receiving endpoint can send the SDP offer in a response if the INVITE did not contain an offer – this is called a Delayed Offer (DO)
• For Early Offer, the answer is sent in a response (usually 200 OK).
• For Delayed Offer, the answer is typically sent in the ACK.
INVITE with SDP - Offer
200 OK with SDP - Answer
Session Established
Phone 1 Unified CM
Early Offer
ACK (no SDP)
BYE
200 OK
INVITE (no SDP)
200 OK with SDP - Offer
Session Established
Phone 1 Unified CM
Delayed Offer
ACK with SDP - Answer
BYE
200 OK
Early Media
• Delayed Offer calls do not set up media until the 200 OK (call is answered)
• If media is required prior to the call being connected, SIP has provisions for Early Media
• With Early Media on a Delayed Offer call, the offer comes from the terminating side in a provisional response (e.g. 183 Session Progress)
• Originating side sends SDP Answer in a PRACK message (defined in RFC 3262)
INVITE (no SDP)
200 OK (INVITE) w/ SDP (should be same as answer)
Session Established
Phone 1 Unified CM
Early Media
ACK
BYE
200 OK
183 Session Progress with SDP - Offer
PRACK with SDP - Answer
Media Stream Established
200 OK (PRACK)
Media Re-negotiation
• Either UA involved in a call can re-INVITE an existing dialog to re-negotiate parameters for the call.
• Cannot re-INVITE until any previous INVITE messages have received a final response.
• UPDATE method can also be used to re-negotiate prior to a final response.
Re-INVITE
Media Re-negotiation
INVITE sip:[email protected]:49833;transport=tls SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901f9c72c19221From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253beDate: Wed, 11 Jan 2012 03:08:51 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM8.6Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 104 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceCall-Info: <urn:x-cisco-remotecc:callinfo>; security= Authenticated; orientation= from; gci= 2-231448; call-instance= 2Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=calling;screen=yes;privacy=offContact: <sip:[email protected]:5061;transport=tls>Content-Type: application/sdpContent-Length: 489
Re-INVITE
Media Re-negotiation
v=0o=CiscoSystemsCCM-SIP 15462272 2 IN IP4 172.18.106.59s=SIP Callc=IN IP4 0.0.0.0t=0 0m=audio 19594 RTP/SAVP 9 101a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXa=rtpmap:9 G722/8000a=ptime:20a=inactivea=rtpmap:101 telephone-event/8000a=fmtp:101 0-15m=video 19444 RTP/AVP 126b=TIAS:1000000a=rtpmap:126 H264/90000a=fmtp:126 profile-level-id=42801E;packetization-mode=1;level-asymmetry-allowed=1a=inactivea=mid:227796888
Re-INVITE – Stopping a Media Session
Media Re-negotiation
INVITE sip:[email protected]:49833;transport=tls SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1bFrom: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253beDate: Wed, 11 Jan 2012 03:08:52 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM8.6Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 106 INVITEMax-Forwards: 70Expires: 180Allow-Events: presenceCall-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= from; gci= 2-231448; call-instance= 2Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=calling;screen=yes;privacy=offContact: <sip:[email protected]:5061;transport=tls>Content-Length: 0
Re-INVITE – Delayed Offer to Re-establish Media Stream
Media Re-negotiation
SIP/2.0 200 OKVia: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fac34c0fb1bFrom: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253beCall-ID: [email protected]: Wed, 11 Jan 2012 03:08:52 GMTCSeq: 106 INVITEServer: Cisco-CPCIUS/9.2.1Contact: <sip:[email protected]:49833;transport=tls>Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFORemote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yesSupported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.2.0,X-cisco-xsi-8.5.1Allow-Events: kpml,dialogRecv-Info: conferenceRecv-Info: x-cisco-conferenceContent-Length: 788Content-Type: application/sdpContent-Disposition: session;handling=optional
Re-INVITE – Offer in 200 OK
Media Re-negotiation
v=0o=Cisco-SIPUA 26259 2 IN IP4 10.116.101.41s=SIP Callt=0 0m=audio 32518 RTP/SAVP 0 8 18 102 9 116 124 101c=IN IP4 10.116.101.41a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXa=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=rtpmap:102 L16/16000a=rtpmap:9 G722/8000a=rtpmap:116 iLBC/8000a=rtpmap:124 ISAC/16000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecvm=video 17614 RTP/AVP 126 97c=IN IP4 10.116.101.41b=TIAS:2500000a=rtpmap:126 H264/90000a=fmtp:126 profile-level-id=42801F;packetization-mode=1;level-asymmetry-allowed=1a=rtpmap:97 H264/90000a=fmtp:97 profile-level-id=42801F;packetization-mode=0;level-asymmetry-allowed=1a=sendrecv
Re-INVITE – Offer in 200 OK
Media Re-negotiation
ACK sip:[email protected]:49833;transport=tls SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK901fb064465a06From: "Paul Giralt" <sip:[email protected]>;tag=15462272~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45545776To: <sip:[email protected]>;tag=0022bdd6843100702aae8e5b-4be253beDate: Wed, 11 Jan 2012 03:08:52 GMTCall-ID: [email protected]: 70CSeq: 106 ACKAllow-Events: presenceContent-Type: application/sdpContent-Length: 446
Re-INVITE – Answer in ACK
Media Re-negotiation
v=0o=CiscoSystemsCCM-SIP 15462272 3 IN IP4 172.18.106.59s=SIP Callt=0 0m=audio 4000 RTP/SAVP 0c=IN IP4 172.18.106.58a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXa=rtpmap:0 PCMU/8000a=ptime:20a=sendonlym=video 0 RTP/AVP 126c=IN IP4 10.116.101.50b=TIAS:1000000a=rtpmap:126 H264/90000a=fmtp:126 profile-level-id=42801E;packetization-mode=1;level-asymmetry-allowed=1a=mid:227796888
Re-INVITE – Answer in ACK – Decline Video Support
DTMF Relay
• 3 Methods for passing DTMF digits over a SIP network:
• RFC 2833
• SIP NOTIFY
• SIP Keypad Markup Language (KPML)
DTMF Relay
• Digits are passed in the RTP stream with a unique payload type
• Capability is negotiated in SDP like any other codec
RFC 2833
m=audio 30414 RTP/AVP 0 8 116 18 100 101
c=IN IP4 172.18.106.231
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/800
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=audio 17236 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Offer Answer
DTMF Relay
• Passes DTMF information in a SIP NOTIFY message telephone-event Event
• Negotiated in Call-Info header
SIP NOTIFY
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK9843c455840434
From: "Paul Giralt" <sip:[email protected]>;tag=14902469~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>
Date: Mon, 13 May 2013 14:48:00 GMT
Call-ID: [email protected]
... snip ...
Call-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
... snip ...
Max-Forwards: 69
Content-Length: 0
Offer
DTMF RelaySIP NOTIFY
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK9843c455840434
From: "Paul Giralt" <sip:[email protected]>;tag=14902469~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>;tag=4363A830-17FC
Call-ID: [email protected]
... snip ...
Allow-Events: telephone-event
Call-Info: <sip:172.18.106.231:5060>;method="NOTIFY;Event=telephone-event;Duration=500”
... snip ...
Content-Length: 601
Answer
DTMF Relay
• Digits passed in payload of a NOTIFY message
SIP NOTIFY
NOTIFY sip:172.18.106.231:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK98443140152a0a
From: "Paul Giralt" <sip:[email protected]>;tag=14902469~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>;tag=4363A830-17FC
Call-ID: [email protected]
CSeq: 104 NOTIFY
Max-Forwards: 70
Date: Mon, 13 May 2013 14:48:11 GMT
User-Agent: Cisco-CUCM10.0
Event: telephone-event
Subscription-State: active
Contact: <sip:172.18.106.59:5060>
P-Asserted-Identity: "Paul Giralt" <sip:[email protected]>
Content-Type: audio/telephone-event
Content-Length: 4
.d
DTMF Relay
• Passes DTMF information in a SIP NOTIFY message kpml Event
• Capability advertised in Allow-Events – uses SUBSCRIBE message to subscribe
SIP KPML
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986efd6c4e51e4
From: "Paul Giralt" <sip:[email protected]>;tag=14918970~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>
Date: Mon, 13 May 2013 15:05:24 GMT
Call-ID: [email protected]
User-Agent: Cisco-CUCM10.0
... snip ...
Allow-Events: presence, kpml
... snip ...
Session-Expires: 18000
Max-Forwards: 69
Content-Length: 0
Offer
DTMF RelaySIP KPML
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986efd6c4e51e4
From: "Paul Giralt" <sip:[email protected]>;tag=14918970~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>;tag=437394E8-2E1
Date: Mon, 13 May 2013 15:05:26 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: kpml, telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Supported: replaces
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Require: timer
Session-Expires: 18000;refresher=uac
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 600
Answer
DTMF RelaySIP KPML
SUBSCRIBE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.231:5060;branch=z9hG4bKBAE27139E
From: <sip:[email protected]>;tag=437394E8-2E1
To: "Paul Giralt" <sip:[email protected]>;tag=14918970~0d0d25d7-4931-4a07-83c6
Call-ID: [email protected]
CSeq: 101 SUBSCRIBE
Max-Forwards: 70
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
Event: kpml
Expires: 7200
Contact: <sip:172.18.106.231:5060>
Content-Type: application/kpml-request+xml
Content-Length: 327
<?xml version="1.0" encoding="UTF-8"?><kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request"
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-
request kpml-request.xsd" version="1.0"><pattern persist="persist"><regex
tag="dtmf">[x*#ABCD]</regex></pattern></kpml-request>
Subscribe to KPML
DTMF RelaySIP KPML
NOTIFY sip:172.18.106.231:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK986f73662cca3b
From: "Paul Giralt" <sip:[email protected]>;tag=14918970~0d0d25d7-4931-4a07-83c6
To: <sip:[email protected]>;tag=437394E8-2E1
Call-ID: [email protected]
CSeq: 104 NOTIFY
Max-Forwards: 70
User-Agent: Cisco-CUCM10.0
Event: kpml
Subscription-State: active;expires=7197
Contact: <sip:[email protected]:5060>
Content-Type: application/kpml-response+xml
Content-Length: 336
<?xml version="1.0" encoding="UTF-8" ?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" xmlns:xsi="http://www.w3.org/2001/XMLSchema-
instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-response kpml-response.xsd" code="200" digits="1"
forced_flush="false" suppressed="false" tag="dtmf" text="Success" version="1.0"/>
Send a Digit
SIP Troubleshooting Tools
• Unified CM / SME Tools:
• Real Time Monitoring Tool / Session Trace
• TranslatorX
• IOS (CUBE) and VCS Troubleshooting Tool:
• TranslatorX
• Wireshark
RTMT Session Trace Tool
• Allows you to search for a call based on calling or called number
• Does not depend on Call Detail Records
• Session trace only traces SIP sessions in detail
• Can display raw SIP messages
• Uses correlation tags to include all call legs related to the call selected
• On versions 8.5 and 8.6, can only be used on calls for which traces still exist on the server. Unified CM 9.0 allows viewing traces that have been archived off-server.
Session Trace Features
TranslatorX Tool
• Parses through Unified CM SDI/SDL Trace Files (and CUBE, CUSP, VCS, Jabber 10.x)
• Drag-and-Drop support for .txt as well as .gz files.
• Latest version supports IOS CUBE ccsip debugs, event-trace, and VCS diagnostic log files
• Decodes SIP, SCCP, H.323, MGCP, Q.Sig, and ISDN Q.931 messages
• Call List based on CDR information in the Traces
• Can generate multi-protocol ladder diagrams
• Sophisticated filtering capabilities
• Download for Windows, Mac OS X, and Linux from:http://translatorx.org/
• NOTE: Do not call TAC for support on TranslatorX (although many TAC engineers use it so feel free to mention you’re using it)
Features
Wireshark
• Open Source network packet capture and analysis tool
• Available at http://www.wireshark.org
• Available for Windows, Mac OS X, and UNIX/Linux
• Provides VoIP Call and SIP analysis
WiresharkHow to Gather a Trace?
• Unified CM, VCS, and IOS provide a mechanism to gather a packet capture
• Will be covered later
Unified CM Trace Configuration
• SIP messaging in Unified CM is written to the SDL trace file when appropriate trace levels are set (SDI trace in for pre-9.0)
• Configured from Cisco Unified Serviceability > Trace > Configuration or by using AnalysisManager
• Unified CM 9.0 combines SDI and SDL traces into the SDL traces
• Unified CM 9.0 fresh install and later default to detailed tracing – no need to configure traces.
Unified CM Trace Configuration
Select the
Server
Select the Service on
Which Trace Needs to
Be Enabled
Select Service
Group
Unified CM Trace Configuration
1. Press
Set Default
2. Set to Detailed
(Should already be Detailed if
running 9.x or later)
Updates All
Servers in This
Cluster with
These Settings
Unified CM Trace Configuration
Enable SIP Stack Trace is NOT needed to see SIP Messages.
Do not enable SIP Stack Trace prior to 9.0 unless directed by TAC.
Okay to have enabled in 9.x and later
Trace Collection
• Various Ways to Collect Trace Files
• RTMT Collect Files
• RTMT Analysis Manager
• RTMT Remote Browse
• RTMT Query Wizard
• OS CLI (file get or file tail)
• file tail activelog cm/trace/ccm/sdl recent
Recommended
Gathering a Packet Capture from Unified CM• Use the Platform CLI command ‘utils network capture’
admin:utils network capture ?
Syntax:
utils network capture [options]
options optional page, numeric, file fname, count num, size bytes, src addr, dest
addr, port num, host protocol addr
admin:utils network capture file capturefile count 100000 size ALL host ip 10.1.1.1
Executing command with options:
size=ALL count=100000 interface=eth0
src= dest= port=
ip=10.1.1.1
admin:file list activelog platform/cli
capturefile.cap
dir count = 0, file count = 1
admin:file get activelog platform/cli/capturefile.cap
Please wait while the system is gathering files info ...done.
Sub-directories were not traversed.
Number of files affected: 1
Total size in Bytes: 24
Total size in Kbytes: 0.0234375
Would you like to proceed [y/n]? y
VCS / Expressway Trace Configuration
• VCS Control / VCS Expressway / Expressway-C / Expressway-E can log SIP messages to a diagnostic log file.
• To enable, navigate to Maintenance > Diagnostics > Diagnostic logging
VCS / Expressway Trace Configuration
Select this if you want
to get a Wireshark
capture
Click Start new log
Click OK to Confirm
CUBE Debugging
• CUBE / IOS Tools:• IOS debugs
• IOS show commands
• Per-call trace
• Event Trace
• Packet export
CUBE Debugging
• When debugging in IOS, configure logging buffered to a fairly large value (based on available memory)
• Disable logging to the console with command ‘no logging console’
• Enable timestamps for debugs
• Make sure router has NTP enabled
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
logging buffered 10000000
no logging console
clock timezone EST -5 0
clock summer-time EDT recurring
ntp server 10.14.1.1
CUBE Debugging
• Various SIP debugs available:
CUBE#debug ccsip ?
all Enable all SIP debugging traces
calls Enable CCSIP SPI calls debugging trace
dhcp Enable SIP-DHCP debugging trace
error Enable SIP error debugging trace
events Enable SIP events debugging trace
function Enable SIP function debugging trace
info Enable SIP info debugging trace
media Enable SIP media debugging trace
messages Enable CCSIP SPI messages debugging trace
preauth Enable SIP preauth debugging traces
states Enable CCSIP SPI states debugging trace
translate Enable SIP translation debugging trace
transport Enable SIP transport debugging traces
verbose Enable verbose mode
CUBE DebuggingSample ‘debug ccsip messages’
Jan 12 03:14:43.102: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 172.18.106.59:5060;branch=z9hG4bK978d2e8df73dc
From: "Paul Giralt" <sip:[email protected]>;tag=16218435~-b82e2c213ca7-45552048
To: <sip:[email protected]>
Date: Thu, 12 Jan 2012 03:09:42 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
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: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3720668288-0000065536-0000015564-0996807340
Session-Expires: 1800
P-Asserted-Identity: "Paul Giralt" <sip:[email protected]>
Remote-Party-ID: "Paul Giralt" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060;transport=tcp>;video;audio
Max-Forwards: 69
Content-Length: 0
CUBE Debugging
• Other generic voice debugs can be useful as well:
• debug voice ccapi inout
• debug voice dialpeer
• debug voice rtp session dtmf-relay
• debug voice rtp session named-event (for any RFC 2833 packets)
Cisco Unified Border Element Basic Call Flow1. Incoming VoIP setup message from originating endpoint
2. This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD, DTMF method, protocol, etc.
3. Match the called number to outbound VoIP dial peer 2
4. Outgoing VoIP setup message
Incoming VoIP Call Outgoing VoIP Call
dial-peer voice 1 voipdestination-pattern 1000incoming called-number .Tsession protocol sipv2session target ipv4:192.168.10.50dtmf-relay rtp-nte sip-kpmlcodec g711ulaw
dial-peer voice 2 voipdestination-pattern 2000session protocol sipv2session target ipv4:192.168.12.25dtmf-relay rtp-ntecodec g711ulaw
Originating Endpoint
TerminatingEndpoint
CUBE
voice service voip
allow-connections sip to sip
CUBE show Commands• show call active voice [brief] shows state of currently active calls
0 : 2807 92135710ms.1 (23:55:20.115 EST Mon Jan 16 2012) +1770 pid:1 Answer 89915644 active
dur 00:00:14 tx:743/14860 rx:718/14360 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.116.101.41:23412 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
0 : 2808 92135720ms.1 (23:55:20.125 EST Mon Jan 16 2012) +1750 pid:100 Originate 9193922900 active
dur 00:00:14 tx:718/14360 rx:755/15100 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 172.30.206.164:10076 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
CUBE show Commands
• show cube calls shows more specific CUBE-related call information and allows filtering
CUBE# show cube calls all ?
callID Display information matches callID
called-number Display information matches called number
calling-number Display information matches calling number
conf-id Display information matches conference ID
detail Display detail level information
fpi-cor Display information matches FPI Correlator
peer-callID Display information matches peer callID
peer-rtp-port Display information matches peer rtp-port number
rtp-port Display information matches rtp-port number
| Output modifiers
CUBE show Commands
CUBE#show cube calls all called-number 8008001180
called number: 8008001180 info are as the following:
=============================================================
=============================================================
Phone number 8008001180 has the following callID associated to it:
=============================================================
CallID: 1730781, calling number: 9195551234
============================================================
A total of 1 call legs associated to number: 8008001180
============================================================
CUBE show CommandsCUBE#show cube calls all callID 1730781
callid: 1730781 info are as the following:
=============================================================
SIP call leg info:
=============================================================
SIP CALL INFO of CCAPI callid 1730781
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 9199915644
Called Number : 8008001180
Called URI : sip:[email protected]:5060
Bit Flags : 0xE04018 0x90000100 0x0
CC Call ID : 1730781
Source IP Address (Sig ): 64.102.250.104
Destn SIP Req Addr:Port : [208.70.21.21]:5060
Destn SIP Resp Addr:Port: [208.70.21.21]:5060
CUBE show CommandsCUBE#show cube calls all callID 1730781
callid: 1730781 info are as the following:
=============================================================
SIP call leg info:
=============================================================
SIP CALL INFO of CCAPI callid 1730781
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 9195551234
Called Number : 8008001180
Called URI : sip:[email protected]:5060
Bit Flags : 0xE04018 0x90000100 0x0
CC Call ID : 1730781
Source IP Address (Sig ): 64.102.250.104
Destn SIP Req Addr:Port : [208.70.21.21]:5060
Destn SIP Resp Addr:Port: [208.70.21.21]:5060
CUBE SIP Event Trace
• All events related to a call are put into buffers
• Can search for calls based on calling number, called number, or call ID
• Buffers can be automatically dumped to an FTP/TFTP server when calls end
• Available on CUBE(Ent) ASR release XE3.10 and later
• Available on CUBE(Ent) ISR release 15.3(3)M and later
1. Define events to be traced
2. Enable event trace (will enable automatically on a reload)
CUBE(config)# monitor event-trace voip ccsip msg size 50
CUBE SIP Event Trace
CUBE# monitor event-trace voip ccsip msg ?
clear Clear the trace
disable Disable tracing
dump Dumps the event buffer into a file
enable Enable tracing
CUBE(config)# monitor event-trace voip ccsip dump-file ftp://user:password@<ipaddr>/path/cube.txt
CUBE(config)# monitor event-trace voip ccsip dump all
3. Configure automatic file uploads (optional)
CUBE SIP Event TraceViewing event trace information
CUBE# monitor event-trace voip ccsip [msg | history] dump filter ? [pretty]
call-id Filter traces based on Internal Call Id
called-num Filter traces based on Called Number
calling-num Filter traces based on Calling Number
sip-call-id Filter traces based on SIP Call Id
CUBE# show monitor event-trace voip ccsip [msg | history] ?
all Show all the traces in current buffer
back Show trace from this far back in the past
clock Show trace from a specific clock time/date
filter Show Trace filter Options
from-boot Show trace from this many seconds after booting
latest Show latest trace events since last display
CUBE SIP Event TraceViewing event trace information
CUBE# show monitor event-trace voip ccsip history filter calling-num 9195551234 latest
--------Cover buff----------
buffer-id = 2828 ccCallId = 305319 PeerCallId = 305320
Called-Number = +18045553456 Calling-Number = 9195551234 Sip-Call-Id =
sip_msgs: Enabled.. Total Traces logged = 18
--------------------------------
--------Cover buff----------
buffer-id = 2829 ccCallId = 305320 PeerCallId = 305319
Called-Number = 8045553456 Calling-Number = 9195551234 Sip-Call-Id = BC182C11-
sip_msgs: Enabled.. Total Traces logged = 12
--------------------------------
CUBE – IP Traffic CaptureExport Packet Data in PCAP Format
IP Traffic Export feature allows export of packets on an interface
Configuration:
• Usage:
ip traffic-export profile CUBE_Debug mode capture
bidirectional
incoming access-list 101
outgoing access-list 101
interface GigabitEthernet0/0
ip traffic-export apply CUBE_Debug size 10000000
traffic-export interface g0/0 start
traffic-export interface g0/0 stop
traffic-export interface g0/0 copy scp://10.1.1.1/capture.pcap
Case Study 1: Unable to Place a CallProblem Description
• A user reports that every time they call (919) 555-1212, they get a message that the call could not be completed as dialed
Case Study 1: Unable to Place a CallUse RTMT Session Trace
• Enter *5551212 into Called Number/URI field
• Set time and duration appropriately
• Search Finds two calls
Case Study 1: Unable to Place a CallUse RTMT Session Trace
• Double-click to see message diagram
• Clearly shows the far-end sends back a 404 Not Found
Case Study 1: Unable to Place a Call
• Enable SIP message debugs – debug ccsip messages
Troubleshoot Call on CUBE
Jan 16 04:00:22.679: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Paul Giralt" <sip:[email protected]>;tag=19210128~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-45568313
To: <sip:[email protected]>
Date: Mon, 16 Jan 2012 03:55:17 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Call-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3852191232-0000065536-0000018595-0996807340
Session-Expires: 1800
P-Asserted-Identity: "Paul Giralt" <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Max-Forwards: 69
Content-Length: 0
Case Study 1: Unable to Place a CallTroubleshoot Call on CUBE
• Check to see if the number matches a valid dial peer
Jan 16 04:00:22.687: //98/E59BC6000000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bKb5291d44b969a4
From: "Paul Giralt" <sip:[email protected]>;tag=19210128~0d0d25d7-4931-4a07-83c6 b82e2c213ca7-45568313
To: <sip:[email protected]>;tag=253488-726
Date: Mon, 16 Jan 2012 04:00:22 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.2.T
Reason: Q.850;cause=1
Content-Length: 0
CUBE#show dialplan number +19195551212
Macro Exp.: +19195551212
No match, result=-1
Case Study 2: Unable to Place a Call #2Problem Description
• A user reports that every time they call another Enterprise number -80010001, they get reorder (fast busy) tone
Case Study 2: Unable to Place a Call #2Use RTMT Session Trace
• Enter *80010001 into Called Number/URI field
• Set time and duration appropriately
• Search Finds one call
Case Study 2: Unable to Place a Call #2Use RTMT Session Trace
• Trace shows signaling from both phone and to destination SIP trunk
• Receiving a 503 Service Unavailable from destination
SIP/2.0 503 Service UnavailableVia: SIP/2.0/UDP 172.18.106.58:5060;branch=z9hG4bK34a0915e2c7a20From: "Paul Giralt" <sip:[email protected]>;tag=5964355~0d0d25d7-4931-4a07-83c6-b82e2c213ca7-44286097To: <sip:[email protected]>;tag=932088316Date: Sun, 07 Jun 2015 16:02:18 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: presenceWarning: 399 collab-ccie-cm1a "Unable to find a device handler for the request received on port 5060 from 172.18.106.58"Content-Length: 0
Case Study 2: Unable to Place a Call #2
Case Study 3: No One Answers the PhoneProblem Description
• A user reports that every time they call a specific phone number, no one answers the call, but if they call from their cell phone, the call is answered immediately every time.
• Calling phone is extension 89919236.
• Called number is 1 877 288-8362
Case Study 3: No One Answers the PhoneCollect Traces
• Problem is reproducible, so generate a test call and then collect traces.
Case Study 3: No One Answers the PhoneUse TranslatorX
• Problem is reproducible, so generate a test call and then collect traces.
• Just drag and drop the folder of traces into the translator window.
Case Study 3: No One Answers the PhoneUse TranslatorX to Analyze Traces
• Search for called party number
Case Study 3: No One Answers the PhoneUse TranslatorX to Analyze Traces
• Disable Filters
• Select the INVITE
• Filter by SIP Call ID (control/command – S)
Case Study 3: No One Answers the PhoneUse TranslatorX to Analyze Traces
03/29/2010 10:36:41.497 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.18.159.231:[5060]:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665
From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543
To: <sip:[email protected]>
Date: Mon, 29 Mar 2010 14:36:41 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.0
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: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 2081204224-3137452793-0000000466-0996807340
Session-Expires: 1800
P-Asserted-Identity: "Test User 1" <sip:[email protected]>
Contact: <sip:[email protected]:5060>;video;audio
Max-Forwards: 69
Content-Length: 0
Case Study 3: No One Answers the PhoneUse TranslatorX to Analyze Traces
• Where did the call originate? Try searching for the calling party number
Case Study 3: No One Answers the PhoneUse TranslatorX to Analyze Traces
• Select the INVITE
• Create New Filter (control/command-N)
• Filter by IP Address (control/command – I)
• Re-enable Filters
Case Study 3: No One Answers the PhoneINVITE from IP Phone w/ SDP
03/29/2010 10:36:33.771 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 1717 bytes:
INVITE sip:[email protected];user=phone SIP/2.0Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0cTo: <sip:[email protected];user=phone>Call-ID: [email protected]: 70Date: Mon, 29 Mar 2010 14:36:33 GMTCSeq: 101 INVITEUser-Agent: Cisco-CP9951/9.0.1Contact: <sip:[email protected]:51682;transport=tls>Expires: 180Accept: application/sdpAllow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFORemote-Party-ID: "Test User 1" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yesSupported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-
control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.0.0,X-cisco-xsi-9.0.1Allow-Events: kpml,dialogContent-Length: 632Content-Type: application/sdpContent-Disposition: session;handling=optional
Case Study 3: No One Answers the PhoneINVITE from IP Phone w/ SDP (continued)
v=0
o=Cisco-SIPUA 26964 0 IN IP4 172.18.159.152
s=SIP Call
t=0 0
m=audio 29254 RTP/SAVP 0 8 18 102 9 116 124 101
c=IN IP4 172.18.159.152
a=crypto:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 25466 RTP/AVP 97
c=IN IP4 172.18.159.152
b=TIAS:1000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801E
a=recvonly
Case Study 3: No One Answers the PhoneUnified CM Sends a 100 Trying
03/29/2010 10:36:33.773 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321 SIP/2.0 100 TryingVia: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0cTo: <sip:[email protected];user=phone>Date: Mon, 29 Mar 2010 14:36:33 GMTCall-ID: [email protected]: 101 INVITEAllow-Events: presenceContent-Length: 0
Case Study 3: No One Answers the PhoneUnified CM Sends a REFER to Play Outside Dialtone
03/29/2010 10:36:33.780 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321 REFER sip:[email protected]:51682 SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151511c5f04bfFrom: <sip:[email protected]>;tag=2144536187To: <sip:[email protected]>Call-ID: [email protected]: 101 REFERMax-Forwards: 70Contact: <sip:[email protected]:5061;transport=tls>User-Agent: Cisco-CUCM8.0Expires: 0Refer-To: cid:[email protected]: <[email protected]>Require: norefersubContent-Type: application/x-cisco-remotecc-request+xmlReferred-By: <sip:[email protected]>Content-Length: 409
Case Study 3: No One Answers the PhoneUnified CM Sends a REFER to play Outside Dialtone (continued)
<x-cisco-remotecc-request><playtonereq><dialogid><callid>[email protected]</callid><localtag>97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542</localtag><remotetag>00260bd9669e07147bcb3aac-3cda8f0c</remotetag>
</dialogid><tonetype>DtOutsideDialTone</tonetype><direction>user</direction>
</playtonereq></x-cisco-remotecc-request>
Case Study 3: No One Answers the PhoneUnified CM Sends a SUBSCRIBE for KPML
03/29/2010 10:36:33.781 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321 SUBSCRIBE sip:[email protected]:51682 SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK1515232b4e84fFrom: <sip:[email protected]>;tag=1976165806To: <sip:[email protected]>Call-ID: [email protected]: 101 SUBSCRIBEDate: Mon, 29 Mar 2010 14:36:33 GMTUser-Agent: Cisco-CUCM8.0Event: kpml; [email protected]; from-tag=00260bd9669e07147bcb3aac-3cda8f0cExpires: 7200Contact: <sip:[email protected]:5061;transport=tls>Accept: application/kpml-response+xmlMax-Forwards: 70Content-Type: application/kpml-request+xmlContent-Length: 424<?xml version="1.0" encoding="UTF-8" ?><kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0"><pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="10000" persist="persist"><regex tag="Backspace OK">[x#*+]|bs</regex>
</pattern></kpml-request>
Case Study 3: No One Answers the PhonePhone Sends 200 OK for the REFER and SUBSCRIBE
03/29/2010 10:36:33.802 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 453 bytes:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151511c5f04bf
From: <sip:[email protected]>;tag=2144536187
To: <sip:[email protected]>;tag=00260bd9669e07167c743311-343ee3af
Call-ID: [email protected]
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 REFER
Server: Cisco-CP9951/9.0.1
Contact: <sip:[email protected]:51682;transport=TLS>
Content-Length: 0
03/29/2010 10:36:33.843 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 465
bytes:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK1515232b4e84f
From: <sip:[email protected]>;tag=1976165806
To: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89
Call-ID: [email protected]
Date: Mon, 29 Mar 2010 14:36:33 GMT
CSeq: 101 SUBSCRIBE
Server: Cisco-CP9951/9.0.1
Contact: <sip:[email protected]:51682;transport=TLS>
Expires: 7200
Content-Length: 0
Case Study 3: No One Answers the PhoneIP Phone
(172.18.159.152)
Unified CM(172.18.159.152)
CUBE(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
Case Study 3: No One Answers the PhoneUser Dials a ‘1’
03/29/2010 10:36:34.350 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 896 bytes:
NOTIFY sip:[email protected]:5061 SIP/2.0Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529baTo: <sip:[email protected]>;tag=1976165806From: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89Call-ID: [email protected]: Mon, 29 Mar 2010 14:36:33 GMTCSeq: 1001 NOTIFYEvent: kpmlSubscription-State: active; expires=7200Max-Forwards: 70Contact: <sip:[email protected]:51682;transport=TLS>Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBEContent-Length: 209Content-Type: application/kpml-response+xmlContent-Disposition: session;handling=required<?xml version="1.0" encoding="UTF-8"?> <kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false"
forced_flush="false" digits="1" tag="Backspace OK"/>
Case Study 3: No One Answers the PhoneUnified CM Replies to NOTIFY With a 200 OK
03/29/2010 10:36:34.352 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321
SIP/2.0 200 OKVia: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529baFrom: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89To: <sip:[email protected]>;tag=1976165806Date: Mon, 29 Mar 2010 14:36:34 GMTCall-ID: [email protected]: 1001 NOTIFYContent-Length: 0
Case Study 3: No One Answers the PhoneUnified CM Replies Sends a REFER to Disable Outside Dialtone
03/29/2010 10:36:34.353 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 REFER sip:[email protected]:51682 SIP/2.0Via: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0From: <sip:[email protected]>;tag=1574166193To: <sip:[email protected]>Call-ID: [email protected]: 101 REFERMax-Forwards: 70Contact: <sip:[email protected]:5061;transport=tls>User-Agent: Cisco-CUCM8.0Expires: 0Refer-To: cid:[email protected]: <[email protected]>Require: norefersubContent-Type: application/x-cisco-remotecc-request+xmlReferred-By: <sip:[email protected]>Content-Length: 401
Case Study 3: No One Answers the Phone<x-cisco-remotecc-request>
<playtonereq>
<dialogid>
<callid>[email protected]</callid>
<localtag>97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542</localtag>
<remotetag>00260bd9669e07147bcb3aac-3cda8f0c</remotetag>
</dialogid>
<tonetype>Dt_NoTone</tonetype>
<direction>user</direction>
</playtonereq>
</x-cisco-remotecc-request>
Case Study 3: No One Answers the PhonePhone Replies With 200 OK to REFER
03/29/2010 10:36:34.402 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 453 bytes:
SIP/2.0 200 OKVia: SIP/2.0/TLS 172.18.106.59:5061;branch=z9hG4bK151536ea86ab0From: <sip:[email protected]>;tag=1574166193To: <sip:[email protected]>;tag=00260bd9669e07184b08b96b-796ab86fCall-ID: [email protected]: Mon, 29 Mar 2010 14:36:33 GMTCSeq: 101 REFERServer: Cisco-CP9951/9.0.1Contact: <sip:[email protected]:51682;transport=TLS>Content-Length: 0
Case Study 3: No One Answers the PhoneIP Phone
(172.18.159.152)
Unified CM(172.18.159.152)
CUBE(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
Case Study 3: No One Answers the PhoneUser Dials a ‘8’
03/29/2010 10:36:34.944 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682 index 2321 with 896 bytes:
NOTIFY sip:[email protected]:5061 SIP/2.0Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK647d03c1To: <sip:[email protected]>;tag=1976165806From: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89Call-ID: [email protected]: Mon, 29 Mar 2010 14:36:34 GMTCSeq: 1002 NOTIFYEvent: kpmlSubscription-State: active; expires=7195Max-Forwards: 70Contact: <sip:[email protected]:51682;transport=TLS>Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBEContent-Length: 209Content-Type: application/kpml-response+xmlContent-Disposition: session;handling=required<?xml version="1.0" encoding="UTF-8"?> <kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false"
forced_flush="false" digits="8" tag="Backspace OK"/>
Case Study 3: No One Answers the PhoneUnified CM Replies to NOTIFY With a 200 OK
03/29/2010 10:36:34.352 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321
SIP/2.0 200 OKVia: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1cd529baFrom: <sip:[email protected]>;tag=00260bd9669e07177ee0d51d-14f56f89To: <sip:[email protected]>;tag=1976165806Date: Mon, 29 Mar 2010 14:36:34 GMTCall-ID: [email protected]: 1001 NOTIFYContent-Length: 0
Case Study 3: No One Answers the PhoneIP Phone
(172.18.159.152)
Unified CM(172.18.159.152)
CUBE(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
Case Study 3: No One Answers the PhoneCUCM Sends an INVITE to the CUBE
03/29/2010 10:36:41.497 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.18.159.231:[5060]:INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543To: <sip:[email protected]>Date: Mon, 29 Mar 2010 14:36:41 GMTCall-ID: [email protected]: timer,resource-priority,replacesMin-SE: 1800User-Agent: Cisco-CUCM8.0Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYCSeq: 101 INVITEExpires: 180Allow-Events: presence, kpmlSupported: X-cisco-srtp-fallbackSupported: GeolocationCall-Info: <sip:172.18.106.59:5060>;method="NOTIFY;Event=telephone-event;Duration=500"Cisco-Guid: 2081204224-3137452793-0000000466-0996807340Session-Expires: 1800P-Asserted-Identity: "Test User 1" <sip:[email protected]>Contact: <sip:[email protected]:5060>;video;audioMax-Forwards: 69Content-Length: 0
Case Study 3: No One Answers the Phone
IP Phone(172.18.159.152)
Unified CM(172.18.159.152)
CUBE(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
INVITE200 OK (SUBSCRIBE)
Case Study 3: No One Answers the PhoneCUBE Replies With a 183 Session Progress W/ SDP
03/29/2010 10:36:42.324 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 1568 from 172.18.159.231:[5060]:SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 172.18.106.59:5060;branch=z9hG4bK1515b3154665From: "Test User 1" <sip:[email protected]>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510543To: <sip:[email protected]>;tag=DE1EFF8-0Date: Mon, 29 Mar 2010 14:37:23 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTERAllow-Events: telephone-eventRemote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=offContact: <sip:[email protected]:5060>Supported: sdp-anatServer: Cisco-SIPGateway/IOS-12.xContent-Type: multipart/mixed;boundary=uniqueBoundaryMime-Version: 1.0Content-Length: 788--uniqueBoundary
Case Study 3: No One Answers the PhoneCUBE Replies With a 183 Session Progress W/ SDP
Content-Type: application/sdpContent-Disposition: session;handling=requiredv=0o=CiscoSystemsSIP-GW-UserAgent 0 7954 IN IP4 172.18.159.231s=SIP Callc=IN IP4 172.18.159.231t=0 0m=audio 27980 RTP/AVP 0 8 116 18 100 101c=IN IP4 172.18.159.231a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:116 iLBC/8000a=fmtp:116 mode=20a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:100 X-NSE/8000a=fmtp:100 192-194a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16--uniqueBoundaryContent-Type: application/x-q931Content-Disposition: signal;handling=optionalContent-Length: 11
Case Study 3: No One Answers the PhoneUnified CM Sends a 180 Ringing to the IP Phone
03/29/2010 10:36:42.330 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682 index 2321
SIP/2.0 180 RingingVia: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0cTo: <sip:[email protected];user=phone>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd-45510542Date: Mon, 29 Mar 2010 14:36:33 GMTCall-ID: [email protected]: 101 INVITEAllow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFYAllow-Events: presenceContact: <sip:[email protected]:5061;transport=tls>Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= to; ui-state= ringout; gci= 2-305505; call-
instance= 1Send-Info: conferenceRemote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=offContent-Length: 0
Case Study 3: No One Answers the PhoneIP Phone
(172.18.159.152)
Unified CM(172.18.159.152)
CUBE(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
INVITE
100 Trying
200 OK (SUBSCRIBE)
180 Ringing
183 Session Progress
Case Study 3: No One Answers the Phone
• Timestamps Jump from 10:36:42 to 10:37:32
• No SIP Signaling for 50 seconds
Phone Keeps Ringing
Case Study 3: No One Answers the PhonePhone Sends a CANCEL
03/29/2010 10:37:32.934 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 172.18.159.152 on port 51682
index 2321 with 422 bytes:
CANCEL sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61
From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0c
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
Max-Forwards: 70
Date: Mon, 29 Mar 2010 14:37:32 GMT
CSeq: 101 CANCEL
User-Agent: Cisco-CP9951/9.0.1
Content-Length: 0
Case Study 3: No One Answers the PhoneUnified CM Sends a 200 OK for the CANCEL
03/29/2010 10:37:32.935 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 172.18.159.152 on port 51682
index 2321
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.18.159.152:51682;branch=z9hG4bK1636ab61
From: "Test User 1" <sip:[email protected]>;tag=00260bd9669e07147bcb3aac-3cda8f0c
To: <sip:[email protected];user=phone>
Date: Mon, 29 Mar 2010 14:37:32 GMT
Call-ID: [email protected]
CSeq: 101 CANCEL
Content-Length: 0
Case Study 3: No One Answers the Phone
IP Phone(172.18.159.152)
Unified CM(172.18.159.152)
CUBE(172.18.159.231)
NOTIFY
200 OK (NOTIFY)
CANCEL
200 OK (CANCEL)
CANCEL
200 OK (CANCEL)
487 Request Cancelled
487 Request Cancelled
ACK
ACK
183 Session Progress (w/ ANSWER)
Phone 1
Case Study 3: No One Answers the Phone
Unified CM
CUBE
SBC (CUBE)SIP
SPSBC
SP SBC
INVITE w/ OFFERINVITE (no SDP)
INVITE (w/ OFFER)
183 Session Progress (w/ OFFER) 180 Ringing (no SDP)
183 Session Progress (w/ ANSWER)
Phone 1
Case Study 3: No One Answers the Phone
Unified CM
CUBE
SBC (CUBE)SIP
SPSBC
SP SBC
INVITE w/ OFFERINVITE (no SDP)
INVITE (w/ OFFER)
183 Session Progress (w/ OFFER)
??? (w/ ANSWER) ??? (w/ ANSWER)
183 Session Progress (w/ ANSWER)
Phone 1
Case Study 3: No One Answers the Phone
Unified CM
CUBE
SBC (CUBE)SIP
SPSBC
SP SBC
INVITE w/ OFFERINVITE (no SDP)
INVITE (w/ OFFER)
183 Session Progress (w/ OFFER)
183 Session Progress (w/ ANSWER) PRACK (w/ ANSWER)
Case Study 3: No One Answers the Phone• How do we get the gateway to cut through audio on the 183 Session Progress message?
• RFC 3262: Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
• Provides a way to acknowledge the 183 Session Progress message – PRACK
• Unified CM parameter “SIP Rel1XX Options” *
•Disabled
•Send PRACK for all 1xx Messages
•Send PRACK if 1xx Contains SDP
*Service Parameter in 7.x and earlier. SIP Profile parameter in 8.x and later
cube(conf-serv-sip)#rel1xx ?
disable Disables reliable-provisional responses
require Requires reliable-provisional responses
supported Supports reliable-provisional responses
Case Study 3: No One Answers the PhoneIP Phone
(172.18.159.152)
Unified CM(172.18.159.152)
CUBE(172.18.159.231)
INVITE
100 Trying
REFER
SUBSCRIBE
200 OK (REFER)
200 OK (SUBSCRIBE)
NOTIFY
200 OK (NOTIFY)
REFER
200 OK (REFER)
NOTIFY
200 OK (NOTIFY)
NOTIFY / 200 OK
Repeats 10 Times
SUBSCRIBE
INVITE
100 Trying
200 OK (SUBSCRIBE)
183 Session Progress
183 Session Progress
PRACK
Case Study 4: Calls to Lync Clients Fail
• When a user dials a Lync client from a video-enabled Cisco 9951 phone, the call fails. Call is from 58574 to 60051.
Case Study 4: Calls to Lync Clients Fail
M = {}
function M.outbound_INVITE(msg)
local contactHeader = msg:getHeader("Contact”)
if contactHeader then
local newContactHeader = string.gsub(contactHeader, ";video;audio;video", "")
msg:modifyHeader("Contact", newContactHeader)
end
end
return M
Case Study 4: Calls to Lync Clients Fail
• For more information:
• Visit http://developer.cisco.com/web/sip/documentation to download the SIP Normalization and Transparency Developer Guide
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