Troubleshooting SIP with Cisco Unified Communications
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- Rolf Richards
- 5 years ago
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2 Troubleshooting SIP with Cisco Unified Communications Paul Giralt, Distinguished Services Engineer
3 Cisco Spark How Questions? Use Cisco Spark to communicate with the speaker after the session 1. Find this session in the Cisco Live Mobile App 2. Click Join the Discussion 3. Install Spark or go directly to the space 4. Enter messages/questions in the space cs.co/ciscolivebot# 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public
4 Agenda Introduction Session Initiation Protocol (SIP) Overview Troubleshooting Tools Unified CM Tracing Expressway / VCS Tracing Cisco Unified Border Element (CUBE) Tracing Sample Call Flows / Case Studies Live Demos
5 SIP Protocol Overview
6 What is SIP? Signaling protocol used to establish, manage, and terminate sessions over an IP network Core protocol defined in RFC 3261 Relies heavily on RFC 3262, RFC 3263, RFC 3264, and RFC 3265 Extended in many, many other RFCs ASCII-based messages 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 6
7 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 7
8 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 8
9 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 9
10 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 10
11 SIP INVITE Method INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1515b From: "Test User 1" To: Call-ID: Supported: timer,resource-priority,replaces User-Agent: Cisco-CUCM12.0 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Session-ID: 52c41df a00074a02fc0cf3b;remote= Supported: Geolocation Call-Info: <sip: :5060>;method="notify;event=telephone-event;duration=500" Cisco-Guid: Session-Expires: 1800 P-Asserted-Identity: "Test User 1" Contact: Max-Forwards: 69 Content-Length: 864 Content-Type: application/sdp 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 11
12 SIP Request Line INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1515b From: "Test User 1" To: Call-ID: URI SIP Version Supported: timer,resource-priority,replaces User-Agent: Cisco-CUCM12.0 Allow: INVITE, SIP Method OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Session-ID: 52c41df a00074a02fc0cf3b;remote= Supported: Geolocation Call-Info: <sip: :5060>;method="notify;event=telephone-event;duration=500" Cisco-Guid: Session-Expires: 1800 P-Asserted-Identity: "Test User 1" Contact: Max-Forwards: 69 Content-Length: 864 Content-Type: application/sdp 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 12
13 SIP Headers INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1515b From: "Test User 1" To: Call-ID: Supported: timer,resource-priority,replaces User-Agent: Cisco-CUCM12.0 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Session-ID: 52c41df a00074a02fc0cf3b;remote= Supported: Geolocation Call-Info: <sip: :5060>;method="notify;event=telephone-event;duration=500" Cisco-Guid: Session-Expires: 1800 P-Asserted-Identity: "Test User 1" Contact: Max-Forwards: 69 Content-Length: 864 Content-Type: application/sdp 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 13
14 SIP Response SIP/ Not Found Via: SIP/2.0/UDP :5060;branch=z9hG4bKb5291d44b969a4 From: "Test User 1" To: Date: Mon, 16 Jan :00:22 GMT Call-ID: CSeq: 101 INVITE Allow-Events: telephone-event Session-ID: 747a0ead a00074a02fc0cf3b;remote=52c41df a00074a02fc0cf3b Server: Cisco-SIPGateway/IOS T Reason: Q.850;cause=1 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 14
15 SIP Response Free-text Reason SIP/ Not Found Via: SIP/2.0/UDP :5060;branch=z9hG4bKb5291d44b969a4 From: "Test User 1" To: Date: Mon, 16 Jan Response :00:22 CodeGMT Call-ID: CSeq: 101 INVITE Allow-Events: telephone-event Session-ID: 747a0ead a00074a02fc0cf3b;remote=52c41df a00074a02fc0cf3b Server: Cisco-SIPGateway/IOS T Reason: Q.850;cause=1 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 15
16 SIP Response SIP/ Not Found Via: SIP/2.0/UDP :5060;branch=z9hG4bKb5291d44b969a4 From: "Test User 1" To: Date: Mon, 16 Jan :00:22 GMT Call-ID: CSeq: 101 INVITE Allow-Events: telephone-event Session-ID: 747a0ead a00074a02fc0cf3b;remote=52c41df a00074a02fc0cf3b Server: Cisco-SIPGateway/IOS T Reason: Q.850;cause=1 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 16
17 SIP Responses Response Code Description Example 1xx 2xx 3xx 4xx Informational Request Received and Continuing to Process Request Success Action was successfully received, understood, and accepted Redirection Another SIP Element needs to be contacted in order to complete the request Client Error Request contains bad syntax or cannot be fulfilled at this server 100 Trying 180 Ringing 183 Session Progress 200 OK 202 Acceptable 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 17
18 Basic SIP Call Setup Phone 1 Unified CM INVITE 200 OK ACK Session Established BYE 200 OK 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 18
19 Basic SIP Call Setup with B2BUA (Unified CM) Phone 1 Unified CM Phone 2 INVITE 200 OK ACK Session Established INVITE 200 OK ACK BYE 200 OK BYE 200 OK 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 19
20 Basic SIP Call Setup with B2BUA (Unified CM) Phone 1 Unified CM SBC (CUBE) INVITE INVITE CUBE 200 OK ACK Session Established 200 OK ACK BYE 200 OK BYE 200 OK 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 19
21 Basic SIP Call Setup with Unified CM and CUBE Phone 1 INVITE Unified CM INVITE SBC (CUBE) CUBE INVITE SBC SP SBC SIP SP 200 OK ACK BYE 200 OK 200 OK ACK Session Established BYE 200 OK 200 OK ACK BYE 200 OK 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 21
22 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 22
23 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 Cisco and/or its affiliates. All rights reserved. Cisco Public 23
24 Session Description Protocol (SDP) - Offer v=0 o=cisco-sipua IN IP s=sip Call t=0 0 m=audio RTP/SAVP c=in IP 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: a=sendrecv m=video RTP/AVP 97 c=in IP b=tias: a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42801e a=recvonly 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 24
25 Session Description Protocol (SDP) - Answer v=0 o=ciscosystemsccm-sip IN IP s=sip Call c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp: m=video 0 RTP/AVP Cisco and/or its affiliates. All rights reserved. Cisco Public 25
26 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 Cisco and/or its affiliates. All rights reserved. Cisco Public 26
27 Early Offer Phone 1 Unified CM INVITE with SDP - Offer 200 OK with SDP - Answer ACK (no SDP) Session Established BYE 200 OK 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 27
28 Delayed Offer Phone 1 Unified CM INVITE (no SDP) 200 OK with SDP - Offer ACK with SDP - Answer Session Established BYE 200 OK 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 28
29 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) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 29
30 Early Media Phone 1 Unified CM INVITE (no SDP) 183 Session Progress with SDP - Offer PRACK with SDP - Answer Media Stream Established 200 OK (PRACK) 200 OK (INVITE) w/ SDP (should be same as answer) ACK Session Established BYE 200 OK 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 30
31 Media Re-negotiation Re-INVITE 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 Cisco and/or its affiliates. All rights reserved. Cisco Public 31
32 Media Re-negotiation Re-INVITE INVITE SIP/2.0 Via: SIP/2.0/TLS :5061;branch=z9hG4bK901f9c72c19221 From: "Paul Giralt" To: Date: Wed, 11 Jan :08:51 GMT Call-ID: Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM12.0 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 104 INVITE Max-Forwards: 70 Expires: 180 Allow-Events: presence Call-Info: <urn:x-cisco-remotecc:callinfo>; security= Authenticated; orientation= from; gci= ; call-instance= 2 Remote-Party-ID: "Paul Giralt" <sip: @ >;party=calling;screen=yes;privacy=off Contact: <sip: @ :5061;transport=tls> Content-Type: application/sdp Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 32
33 Media Re-negotiation Re-INVITE Stopping a Media Session v=0 o=ciscosystemsccm-sip IN IP s=sip Call c=in IP t=0 0 m=audio RTP/SAVP a=crypto:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx a=rtpmap:9 G722/8000 a=ptime:20 a=inactive a=rtpmap:101 telephone-event/8000 a=fmtp: m=video RTP/AVP 126 b=tias: a=rtpmap:126 H264/90000 a=fmtp:126 profile-level-id=42801e;packetization-mode=1;level-asymmetry-allowed=1 a=inactive a=mid: Cisco and/or its affiliates. All rights reserved. Cisco Public 33
34 Media Re-negotiation Re-INVITE Delayed Offer to Re-establish Media Stream INVITE SIP/2.0 Via: SIP/2.0/TLS :5061;branch=z9hG4bK901fac34c0fb1b From: "Paul Giralt" To: Date: Wed, 11 Jan :08:52 GMT Call-ID: Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM12.0 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 106 INVITE Max-Forwards: 70 Expires: 180 Allow-Events: presence Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= from; gci= ; call-instance= 2 Remote-Party-ID: "Paul Giralt" <sip: @ >;party=calling;screen=yes;privacy=off Contact: <sip: @ :5061;transport=tls> Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 34
35 Media Re-negotiation Re-INVITE Offer in 200 OK SIP/ OK Via: SIP/2.0/TLS :5061;branch=z9hG4bK901fac34c0fb1b From: "Paul Giralt" To: Call-ID: Date: Wed, 11 Jan :08:52 GMT CSeq: 106 INVITE Server: Cisco-8865/ Contact: Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO Remote-Party-ID: "Paul Giralt" Supported: replaces,join,sdp-anat,norefersub,extended-refer,x-cisco-callinfo,x-cisco-serviceuri,x-cisco-escapecodes,x-cisco-servicecontrol,x-cisco-srtp-fallback,x-cisco-monrec,x-cisco-config,x-cisco-sis-5.2.0,x-cisco-xsi Allow-Events: kpml,dialog Recv-Info: conference Recv-Info: x-cisco-conference Content-Length: 788 Content-Type: application/sdp Content-Disposition: session;handling=optional 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 35
36 Media Re-negotiation Re-INVITE Offer in 200 OK v=0 o=cisco-sipua IN IP s=sip Call t=0 0 m=audio RTP/SAVP c=in IP a=crypto:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:102 L16/16000 a=rtpmap:9 G722/8000 a=rtpmap:116 ilbc/8000 a=rtpmap:124 ISAC/16000 a=rtpmap:101 telephone-event/8000 a=fmtp: a=sendrecv m=video RTP/AVP c=in IP b=tias: a=rtpmap:126 H264/90000 a=fmtp:126 profile-level-id=42801f;packetization-mode=1;level-asymmetry-allowed=1 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42801f;packetization-mode=0;level-asymmetry-allowed=1 a=sendrecv 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 36
37 Media Re-negotiation Re-INVITE Answer in ACK ACK SIP/2.0 Via: SIP/2.0/TLS :5061;branch=z9hG4bK901fb064465a06 From: "Paul Giralt" To: Date: Wed, 11 Jan :08:52 GMT Call-ID: Max-Forwards: 70 CSeq: 106 ACK Allow-Events: presence Content-Type: application/sdp Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 37
38 Media Re-negotiation Re-INVITE Answer in ACK Decline Video Support v=0 o=ciscosystemsccm-sip IN IP s=sip Call t=0 0 m=audio 4000 RTP/SAVP 0 c=in IP a=crypto:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendonly m=video 0 RTP/AVP 126 c=in IP b=tias: a=rtpmap:126 H264/90000 a=fmtp:126 profile-level-id=42801e;packetization-mode=1;level-asymmetry-allowed=1 a=mid: Cisco and/or its affiliates. All rights reserved. Cisco Public 38
39 DTMF Relay 3 Methods for passing DTMF digits over a SIP network: RFC 4933 (a.k.a. RFC 2833) SIP NOTIFY SIP Keypad Markup Language (KPML) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 39
40 DTMF Relay RFC 4933 / RFC 2833 Digits are passed in the RTP stream with a unique payload type Capability is negotiated in SDP like any other codec Offer m=audio RTP/AVP c=in IP 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: a=rtpmap:101 telephone-event/8000 a=fmtp: Answer m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp: Cisco and/or its affiliates. All rights reserved. Cisco Public 40
41 DTMF Relay SIP NOTIFY Passes DTMF information in a SIP NOTIFY message telephone-event Event Negotiated in Call-Info header Offer INVITE sip: @ :5060 SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK9843c From: "Paul Giralt" <sip: @ >;tag= ~0d0d25d a07-83c6 To: <sip: @ > Date: Mon, 13 May :48:00 GMT Call-ID: 1a fd20-962c99-3b6a12ac@ snip... Call-Info: <sip: :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: Cisco and/or its affiliates. All rights reserved. Cisco Public 41
42 DTMF Relay SIP NOTIFY Answer SIP/ OK Via: SIP/2.0/UDP :5060;branch=z9hG4bK9843c From: "Paul Giralt" To: Call-ID: snip... Allow-Events: telephone-event Call-Info: <sip: :5060>;method="notify;event=telephone-event;duration= snip... Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 42
43 DTMF Relay SIP NOTIFY Digits passed in payload of a NOTIFY message NOTIFY sip: :5060 SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK a0a From: "Paul Giralt" <sip: @ >;tag= ~0d0d25d a07-83c6 To: <sip: @ >;tag=4363a830-17fc Call-ID: 1a fd20-962c99-3b6a12ac@ CSeq: 104 NOTIFY Max-Forwards: 70 Date: Mon, 13 May :48:11 GMT User-Agent: Cisco-CUCM12.0 Event: telephone-event Subscription-State: active Contact: <sip: :5060> P-Asserted-Identity: "Paul Giralt" <sip: @ > Content-Type: audio/telephone-event Content-Length: 4.d 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 43
44 DTMF Relay SIP KPML Passes DTMF information in a SIP NOTIFY message kpml Event Capability advertised in Allow-Events uses SUBSCRIBE message to subscribe Offer INVITE sip: @ :5060 SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK986efd6c4e51e4 From: "Paul Giralt" <sip: @ >;tag= ~0d0d25d a07-83c6 To: <sip: @ > Date: Mon, 13 May :05:24 GMT Call-ID: 885e f-3b6a12ac@ User-Agent: Cisco-CUCM snip... Allow-Events: presence, kpml... snip... Session-Expires: Max-Forwards: 69 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 44
45 DTMF Relay SIP KPML Answer SIP/ OK Via: SIP/2.0/UDP :5060;branch=z9hG4bK986efd6c4e51e4 From: "Paul Giralt" To: Date: Mon, 13 May :05:26 GMT Call-ID: CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: kpml, telephone-event Remote-Party-ID: Contact: Supported: replaces Server: Cisco-SIPGateway/IOS T Require: timer Session-Expires: 18000;refresher=uac Content-Type: multipart/mixed;boundary=uniqueboundary Mime-Version: 1.0 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 45
46 DTMF Relay SIP KPML Subscribe to KPML SUBSCRIBE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bKBAE27139E From: To: "Paul Giralt" Call-ID: CSeq: 101 SUBSCRIBE Max-Forwards: 70 User-Agent: Cisco-SIPGateway/IOS T Event: kpml Expires: 7200 Contact: <sip: :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=" 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></kpmlrequest> 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 46
47 DTMF Relay SIP KPML Send a Digit NOTIFY sip: :5060 SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK986f73662cca3b From: "Paul Giralt" <sip: @ >;tag= ~0d0d25d a07-83c6 To: <sip: @ >;tag=437394e8-2e1 Call-ID: 885e f-3b6a12ac@ CSeq: 104 NOTIFY Max-Forwards: 70 User-Agent: Cisco-CUCM12.0 Event: kpml Subscription-State: active;expires=7197 Contact: <sip: @ :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=" 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"/> 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 47
48 SIP Session-ID (RFC 7989) Session-ID Header carries an end-to-end session identifier Allows you to track a call as it traverses through various SIP systems Session-ID: 65596d a00074a02fc0d796;remote=747a0ead a00074a02fc0cf3b Local UUID Remote UUID Session Identifier represented as {A,B} referring to {local,remote} UUIDs {A,B} is equivalent to {B,A} 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 48
49 SIP Call with Session-ID Headers Phone 1 Unified CM Phone 2 Phone 3 INVITE {A,N} 100 Trying {N,A} 200 OK {B,A} ACK {A,B} INVITE {A,N} 100 Trying {B,A} 200 OK {B,A} ACK {A,B} Session Established REFER {B,A} 200 OK {A,B} 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 49
50 SIP Call with Session-ID Headers Phone 1 Unified CM Phone 2 Phone 3 INVITE {A,N} re-invite {C,A} 200 OK {A,C} 200 OK {C,A} ACK {A,C} Session Established BYE {A,B} 200 OK {B,A} 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 50
51 SIP Session-ID Support Current Support: Unified CM 11.0 and later Jabber 11.5 and later 78XX/88XX Endpoints 11.0 and later CUBE 15.6(2)T CVP 11.5 VCS / Expressway X8.10 Cisco Spark Planned Upcoming Support DX & MX-series on CE software 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 51
52 Troubleshooting Tools
53 SIP Troubleshooting Tools Unified CM / SME Tools: Real Time Monitoring Tool / Session Trace TranslatorX IOS (CUBE) and VCS Troubleshooting Tool: TranslatorX Wireshark 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 53
54 RTMT Session Trace Tool Session Trace Features Allows you to search for a call based on calling or called number Does not depend on Call Detail Records (uses calllogs file) Session trace only traces SIP sessions (no SCCP, H.323, or MGCP) Can display raw SIP messages Uses correlation tags to include all call legs related to the call selected 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 54
55 RTMT Session Trace Tool 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 55
56 RTMT Session Trace Tool Call Flow Diagram 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 56
57 RTMT Session Trace Tool Click on the message in the call flow diagram to see the actual message 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 57
58 TranslatorX Tool Features Parses through Unified CM SDI/SDL Trace Files (and CUBE, CUSP, VCS/Expressway, Jabber 10.x+) Drag-and-Drop support for.txt as well as.gz 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 (including Session-ID Support) Download for Windows, Mac OS X, and Linux from: NOTE: Do not call TAC for support on TranslatorX (although many TAC engineers use it so feel free to mention you re using it) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 58
59 TranslatorX Tool 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 59
60 TranslatorX Tool Call List Window 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 60
61 TranslatorX Tool Call List Filtering Double-click for complete Call Detail Record 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 61
62 TranslatorX Tool CDR View 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 62
63 TranslatorX Tool CDR View 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 63
64 TranslatorX Tool Generating Filters Select a Call and click Generate Filter button 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 64
65 TranslatorX Tool - Filters 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 65
66 TranslatorX Tool Call Flow Diagram 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 66
67 TranslatorX Tool Call Flow Diagram 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 67
68 Wireshark Open Source network packet capture and analysis tool Available at Available for Windows, Mac OS X, and UNIX/Linux Provides VoIP Call and SIP analysis 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 68
69 Wireshark 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 69
70 Wireshark VoIP Call Analysis 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 70
71 Wireshark VoIP Call Ladder Diagram 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 71
72 Wireshark How to Gather a Trace? Unified CM, VCS, and IOS provide a mechanism to gather a packet capture Will be covered later 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 72
73 Unified CM Trace Configuration
74 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 Analysis Manager Unified CM 9.0 and later combines SDI and SDL traces into the SDL traces Unified CM 9.0 and later fresh install and later default to detailed tracing no need to configure traces Cisco and/or its affiliates. All rights reserved. Cisco Public 74
75 Unified CM Trace Configuration Select the Server Select Service Group Select the Service on Which Trace Needs to Be Enabled 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 75
76 Unified CM Trace Configuration 1. Press Set Default Updates All Servers in this cluster with these settings 2. Set to Detailed 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 76
77 Unified CM Trace Configuration 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 77
78 Trace Collection Various Ways to Collect Trace Files RTMT Collect Files RTMT Analysis Manager RTMT Remote Browse RTMT Query Wizard Recommended OS CLI (file get or file tail) file tail activelog cm/trace/ccm/sdl recent 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 78
79 Gathering a Packet Capture from Unified CM Use the Platform CLI command utils network capture Reference 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 size ALL host ip Executing command with options: size=all count= interface=eth0 src= dest= port= ip= 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: Would you like to proceed [y/n]? y 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 79
80 Expressway / VCS Trace Configuration
81 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 81
82 VCS / Expressway Trace Configuration Select this if you want to get a Wireshark capture Click Start new log Click OK to Confirm 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 82
83 VCS / Expressway Trace Configuration Click Stop logging after reproducing your problem 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 83
84 VCS / Expressway Trace Configuration Click to Download the log and tcpdump (for Wireshark) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 84
85 Cisco Unified Border Element (CUBE) Trace Configuration
86 CUBE Debugging CUBE / IOS Tools: IOS debugs IOS show commands Event Trace Packet export 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 86
87 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 no logging console clock timezone EST -5 0 clock summer-time EDT recurring ntp server Cisco and/or its affiliates. All rights reserved. Cisco Public 87
88 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 non-call Enable Non-Call-Context trace (OPTIONS, SUBSCRIBE etc) 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 88
89 CUBE Debugging Sample debug ccsip messages Jan 12 03:14:43.102: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE SIP/2.0 Via: SIP/2.0/TCP :5060;branch=z9hG4bK978d2e8df73dc From: "Paul Giralt" To: Date: Thu, 12 Jan :09:42 GMT Call-ID: Supported: 100rel,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 Supported: X-cisco-srtp-fallback Supported: Geolocation Call-Info: <sip: :5060>;method="notify;event=telephone-event;duration=500" Cisco-Guid: Session-Expires: 1800 P-Asserted-Identity: "Paul Giralt" Remote-Party-ID: "Paul Giralt" Contact: Max-Forwards: 69 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 89
90 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 4933/2833 packets) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 90
91 Cisco Unified Border Element Basic Call Flow Incoming VoIP setup message from originating endpoint This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD, DTMF method, protocol, etc. Match the called number to outbound VoIP dial peer 2 Outgoing VoIP setup message voice service voip allow-connections sip to sip Originating Endpoint Incoming VoIP Call Outgoing VoIP Call Terminating Endpoint CUBE dial-peer voice 1 voip destination-pattern 1000 incoming called-number.t session protocol sipv2 session target ipv4: dtmf-relay rtp-nte sip-kpml codec g711ulaw dial-peer voice 2 voip destination-pattern 2000 session protocol sipv2 session target ipv4: voice-class sip early-offer forced dtmf-relay rtp-nte codec g711ulaw 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 91
92 CUBE show Commands show call active voice [brief] shows state of currently active calls 0 : ms.1 (23:55: EST Mon Jan ) pid:1 Answer active dur 00:00:14 tx:743/14860 rx:718/14360 dscp:0 media:0 audio tos:0xb8 video tos:0x0 IP :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 : ms.1 (23:55: EST Mon Jan ) pid:100 Originate active dur 00:00:14 tx:718/14360 rx:755/15100 dscp:0 media:0 audio tos:0xb8 video tos:0x0 IP :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: Cisco and/or its affiliates. All rights reserved. Cisco Public 92
93 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 93
94 CUBE show Commands CUBE#show cube calls all called-number called number: info are as the following: ============================================================= ============================================================= Phone number has the following callid associated to it: ============================================================= CallID: , calling number: ============================================================ A total of 1 call legs associated to number: ============================================================ 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 94
95 CUBE show Commands CUBE#show cube calls all callid callid: info are as the following: ============================================================= SIP call leg info: ============================================================= SIP CALL INFO of CCAPI callid Call 1 SIP Call ID : 41669A03-D5611E5-8A6384B0-5DED4CDB@ State of the call : STATE_ACTIVE (7) Substate of the call : SUBSTATE_NONE (0) Calling Number : Called Number : Called URI : sip: @ :5060 Bit Flags : 0xE x x0 CC Call ID : Source IP Address (Sig ): Destn SIP Req Addr:Port : [ ]:5060 Destn SIP Resp Addr:Port: [ ]: Cisco and/or its affiliates. All rights reserved. Cisco Public 95
96 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 96
97 Reference CUBE SIP Event Trace 1. Define events to be traced CUBE(config)# monitor event-trace voip ccsip msg size Enable event trace (will enable automatically on a reload) 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 3. Configure automatic file uploads (optional) 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 97
98 CUBE SIP Event Trace Viewing event trace information Reference 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 98
99 CUBE SIP Event Trace Viewing event trace information Reference CUBE# show monitor event-trace voip ccsip history filter calling-num latest Cover buff buffer-id = 2828 cccallid = PeerCallId = Called-Number = Calling-Number = Sip-Call-Id = e012f98-167c899-3b6a12ac@ sip_msgs: Enabled.. Total Traces logged = Cover buff buffer-id = 2829 cccallid = PeerCallId = Called-Number = Calling-Number = Sip-Call-Id = BC182C11-82D611E3-BAACD07D-93FD9A72@ sip_msgs: Enabled.. Total Traces logged = Cisco and/or its affiliates. All rights reserved. Cisco Public 99
100 CUBE IP Traffic Capture Export Packet Data in PCAP Format IP Traffic Export feature allows export of packets on an interface Reference 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 traffic-export interface g0/0 start traffic-export interface g0/0 stop traffic-export interface g0/0 copy scp:// /capture.pcap 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 100
101 Case Studies
102 Case Study 1: Unable to Place a Call Problem Description A user reports that every time they call (919) , they get a message that the call could not be completed as dialed 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 102
103 Case Study 1: Unable to Place a Call Use RTMT Session Trace Enter * into Called Number/URI field Set time and duration appropriately Search Finds two calls 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 103
104 Case Study 1: Unable to Place a Call Use RTMT Session Trace Double-click to see message diagram Clearly shows the farend sends back a 404 Not Found 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 104
105 Case Study 1: Unable to Place a Call Troubleshoot Call on CUBE Enable SIP message debugs debug ccsip messages Jan 16 04:00:22.679: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip: @ :5060 SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bKb5291d44b969a4 From: "Paul Giralt" <sip: @ >;tag= ~0d0d25d a07-83c6-b82e2c213ca To: <sip: @ > Date: Mon, 16 Jan :55:17 GMT Call-ID: e59bc600-f1319fa5-b1ea4a-3b6a12ac@ Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM12.0 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Call-Info: <sip: :5060>;method="notify;event=telephone-event;duration=500" Cisco-Guid: Session-Expires: 1800 P-Asserted-Identity: "Paul Giralt" <sip: @ > Contact: <sip: @ :5060> Max-Forwards: 69 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 105
106 Case Study 1: Unable to Place a Call Troubleshoot Call on CUBE Check to see if the number matches a valid dial peer Jan 16 04:00:22.687: //98/E59BC /SIP/Msg/ccsipDisplayMsg: Sent: SIP/ Not Found Via: SIP/2.0/UDP :5060;branch=z9hG4bKb5291d44b969a4 From: "Paul Giralt" <sip: @ >;tag= ~0d0d25d a07-83c6 b82e2c213ca To: <sip: @ >;tag= Date: Mon, 16 Jan :00:22 GMT Call-ID: e59bc600-f1319fa5-b1ea4a-3b6a12ac@ CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS T Reason: Q.850;cause=1 Content-Length: 0 CUBE#show dialplan number Macro Exp.: No match, result= Cisco and/or its affiliates. All rights reserved. Cisco Public 106
107 Case Study 2: Unable to Place a Call #2 Problem Description A user reports that every time they call , they get reorder (fast busy) tone 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 107
108 Case Study 2: Unable to Place a Call #2 Use RTMT Session Trace Enter * into Called Number/URI field Set time and duration appropriately Search Finds one call 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 108
109 Case Study 2: Unable to Place a Call #2 Use RTMT Session Trace Trace shows signaling from both phone and to destination SIP trunk Receiving a 503 Service Unavailable from destination 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 109
110 Case Study 2: Unable to Place a Call #2 SIP/ Service Unavailable Via: SIP/2.0/UDP :5060;branch=z9hG4bK34a0915e2c7a20 From: "Paul Giralt" <sip: @ >;tag= ~0d0d25d a07-83c6-b82e2c213ca To: <sip: @ >;tag= Date: Sun, 07 Jun :02:18 GMT Call-ID: 9126a b0a-349ed0-3a6a12ac@ CSeq: 101 INVITE Allow-Events: presence Warning: 399 collab-ccie-cm1a "Unable to find a device handler for the request received on port 5060 from " Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 110
111 Case Study 3: No One Answers the Phone Problem 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 Called number is Cisco and/or its affiliates. All rights reserved. Cisco Public 111
112 Case Study 3: No One Answers the Phone Collect Traces Problem is reproducible, so generate a test call and then collect traces Cisco and/or its affiliates. All rights reserved. Cisco Public 112
113 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces Problem is reproducible, so generate a test call and then collect traces. Drag and Drop folder into TranslatorX 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 113
114 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces Try to find call in Call List 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 114
115 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces Search for called party number 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 115
116 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces Disable Filters Select the INVITE Filter by SIP Call ID (control/command S) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 116
117 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces 03/29/ :36: //SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to :[5060]: INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1515b From: "Test User 1" To: Date: Mon, 29 Mar :36:41 GMT Call-ID: 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: :5060>;method="notify;event=telephone-event;duration=500" Cisco-Guid: Session-Expires: 1800 P-Asserted-Identity: "Test User 1" Contact: Max-Forwards: 69 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 117
118 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces Where did the call originate? Try searching for the calling party number 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 118
119 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces Select the INVITE Create New Filter (control/command-n) Filter by IP Address (control/command I) Re-enable Filters 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 119
120 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 120
121 Case Study 3: No One Answers the Phone INVITE from IP Phone w/ SDP 03/29/ :36: //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from on port index 2321 with 1717 bytes: INVITE SIP/2.0 Via: SIP/2.0/TLS :51682;branch=z9hG4bK1636ab61 From: "Test User 1" To: Call-ID: Max-Forwards: 70 Date: Mon, 29 Mar :36:33 GMT CSeq: 101 INVITE User-Agent: Cisco-CP9951/9.0.1 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO Remote-Party-ID: "Test User 1" Supported: replaces,join,sdp-anat,norefersub,extended-refer,x-cisco-callinfo,x-cisco-serviceuri,x-cisco-escapecodes,x-ciscoservice-control,x-cisco-srtp-fallback,x-cisco-monrec,x-cisco-config,x-cisco-sis-5.0.0,x-cisco-xsi Allow-Events: kpml,dialog Content-Length: 632 Content-Type: application/sdp Content-Disposition: session;handling=optional 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 121
122 Case Study 3: No One Answers the Phone INVITE from IP Phone w/ SDP (continued) v=0 o=cisco-sipua IN IP s=sip Call t=0 0 m=audio RTP/SAVP c=in IP 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: a=sendrecv m=video RTP/AVP 97 c=in IP b=tias: a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42801e a=recvonly 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 122
123 Case Study 3: No One Answers the Phone Unified CM Sends a 100 Trying 03/29/ :36: //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to on port index 2321 SIP/ Trying Via: SIP/2.0/TLS :51682;branch=z9hG4bK1636ab61 From: "Test User 1" <sip: @ >;tag=00260bd9669e07147bcb3aac-3cda8f0c To: <sip:9@ ;user=phone> Date: Mon, 29 Mar :36:33 GMT Call-ID: 00260bd9-669e000b-588c0c2b-2193e2a3@ CSeq: 101 INVITE Allow-Events: presence Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 123
124 Case Study 3: No One Answers the Phone Unified CM Sends a REFER to Play Outside Dialtone 03/29/ :36: //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to on port index 2321 REFER sip: @ :51682 SIP/2.0 Via: SIP/2.0/TLS :5061;branch=z9hG4bK151511c5f04bf From: <sip: @ >;tag= To: <sip: @ > Call-ID: 7747f400-bb01baf b6a12ac@ CSeq: 101 REFER Max-Forwards: 70 Contact: <sip: @ :5061;transport=tls> User-Agent: Cisco-CUCM8.0 Expires: 0 Refer-To: cid: @ Content-Id: < @ > Require: norefersub Content-Type: application/x-cisco-remotecc-request+xml Referred-By: <sip: @ > Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 124
125 Case Study 3: No One Answers the Phone Unified CM Sends a REFER to play Outside Dialtone (continued) <x-cisco-remotecc-request> <playtonereq> <dialogid> <callid>00260bd9-669e000b-588c0c2b-2193e2a3@ </callid> <localtag>97903bc0-a3de-4a15-ba27-44c81fe3adcd </localtag> <remotetag>00260bd9669e07147bcb3aac-3cda8f0c</remotetag> </dialogid> <tonetype>dtoutsidedialtone</tonetype> <direction>user</direction> </playtonereq> </x-cisco-remotecc-request> 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 125
126 Case Study 3: No One Answers the Phone Unified CM Sends a SUBSCRIBE for KPML 03/29/ :36: //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to on port index 2321 SUBSCRIBE sip: @ :51682 SIP/2.0 Via: SIP/2.0/TLS :5061;branch=z9hG4bK b4e84f From: <sip:9@ >;tag= To: <sip: @ > Call-ID: 7747f400-bb01baf b6a12ac@ CSeq: 101 SUBSCRIBE Date: Mon, 29 Mar :36:33 GMT User-Agent: Cisco-CUCM8.0 Event: kpml; call-id=00260bd9-669e000b-588c0c2b-2193e2a3@ ; from-tag=00260bd9669e07147bcb3aac-3cda8f0c Expires: 7200 Contact: <sip:9@ :5061;transport=tls> Accept: application/kpml-response+xml Max-Forwards: 70 Content-Type: application/kpml-request+xml Content-Length: 424 <?xml version="1.0" encoding="utf-8"?> <kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi=" 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> 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 126
127 Case Study 3: No One Answers the Phone Phone Sends 200 OK for the REFER and SUBSCRIBE 03/29/ :36: //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from on port index 2321 with 453 bytes: SIP/ OK Via: SIP/2.0/TLS :5061;branch=z9hG4bK151511c5f04bf From: To: Call-ID: Date: Mon, 29 Mar :36:33 GMT CSeq: 101 REFER Server: Cisco-CP9951/9.0.1 Contact: Content-Length: 0 03/29/ :36: //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from on port index 2321 with 465 bytes: SIP/ OK Via: SIP/2.0/TLS :5061;branch=z9hG4bK b4e84f From: <sip:9@ >;tag= To: <sip: @ >;tag=00260bd9669e07177ee0d51d-14f56f89 Call-ID: 7747f400-bb01baf b6a12ac@ Date: Mon, 29 Mar :36:33 GMT CSeq: 101 SUBSCRIBE Server: Cisco-CP9951/9.0.1 Contact: <sip:4a8a8f91-609e-d655-19ea-44eedcd7b0d6@ :51682;transport=tls> Expires: 7200 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 127
128 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) CUBE ( ) INVITE 100 Trying REFER SUBSCRIBE 200 OK (REFER) 200 OK (SUBSCRIBE) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 128
129 Case Study 3: No One Answers the Phone User Dials a 1 03/29/ :36: //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from on port index 2321 with 896 bytes: NOTIFY sip:9@ :5061 SIP/2.0 Via: SIP/2.0/TLS :51682;branch=z9hG4bK1cd529ba To: <sip:9@ >;tag= From: <sip: @ >;tag=00260bd9669e07177ee0d51d-14f56f89 Call-ID: 7747f400-bb01baf b6a12ac@ Date: Mon, 29 Mar :36:33 GMT CSeq: 1001 NOTIFY Event: kpml Subscription-State: active; expires=7200 Max-Forwards: 70 Contact: <sip:4a8a8f91-609e-d655-19ea-44eedcd7b0d6@ :51682;transport=tls> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Content-Length: 209 Content-Type: application/kpml-response+xml Content-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"/> 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 129
130 Case Study 3: No One Answers the Phone Unified CM Replies to NOTIFY With a 200 OK 03/29/ :36: //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to on port index 2321 SIP/ OK Via: SIP/2.0/TLS :51682;branch=z9hG4bK1cd529ba From: <sip: @ >;tag=00260bd9669e07177ee0d51d-14f56f89 To: <sip:9@ >;tag= Date: Mon, 29 Mar :36:34 GMT Call-ID: 7747f400-bb01baf b6a12ac@ CSeq: 1001 NOTIFY Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 130
131 Case Study 3: No One Answers the Phone Unified CM Replies Sends a REFER to Disable Outside Dialtone 03/29/ :36: //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to on port REFER sip: @ :51682 SIP/2.0 Via: SIP/2.0/TLS :5061;branch=z9hG4bK151536ea86ab0 From: <sip: @ >;tag= To: <sip: @ > Call-ID: 77e08a80-bb01baf b6a12ac@ CSeq: 101 REFER Max-Forwards: 70 Contact: <sip: @ :5061;transport=tls> User-Agent: Cisco-CUCM8.0 Expires: 0 Refer-To: cid: @ Content-Id: < @ > Require: norefersub Content-Type: application/x-cisco-remotecc-request+xml Referred-By: <sip: @ > Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 131
132 Case Study 3: No One Answers the Phone <x-cisco-remotecc-request> <playtonereq> <dialogid> <localtag>97903bc0-a3de-4a15-ba27-44c81fe3adcd </localtag> <remotetag>00260bd9669e07147bcb3aac-3cda8f0c</remotetag> </dialogid> <tonetype>dt_notone</tonetype> <direction>user</direction> </playtonereq> </x-cisco-remotecc-request> 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 132
133 Case Study 3: No One Answers the Phone Phone Replies With 200 OK to REFER 03/29/ :36: //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from on port index 2321 with 453 bytes: SIP/ OK Via: SIP/2.0/TLS :5061;branch=z9hG4bK151536ea86ab0 From: To: Call-ID: Date: Mon, 29 Mar :36:33 GMT CSeq: 101 REFER Server: Cisco-CP9951/9.0.1 Contact: Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 133
134 Case Study 3: No One Answers the Phone IP Phone ( ) INVITE 100 Trying REFER SUBSCRIBE 200 OK (REFER) 200 OK (SUBSCRIBE) NOTIFY 200 OK (NOTIFY) REFER 200 OK (REFER) Unified CM ( ) CUBE ( ) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 134
135 Case Study 3: No One Answers the Phone User Dials a 8 03/29/ :36: //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from on port index 2321 with 896 bytes: NOTIFY sip:9@ :5061 SIP/2.0 Via: SIP/2.0/TLS :51682;branch=z9hG4bK647d03c1 To: <sip:9@ >;tag= From: <sip: @ >;tag=00260bd9669e07177ee0d51d-14f56f89 Call-ID: 7747f400-bb01baf b6a12ac@ Date: Mon, 29 Mar :36:34 GMT CSeq: 1002 NOTIFY Event: kpml Subscription-State: active; expires=7195 Max-Forwards: 70 Contact: <sip:4a8a8f91-609e-d655-19ea-44eedcd7b0d6@ :51682;transport=tls> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Content-Length: 209 Content-Type: application/kpml-response+xml Content-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"/> 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 135
136 Case Study 3: No One Answers the Phone Unified CM Replies to NOTIFY With a 200 OK 03/29/ :36: //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to on port index 2321 SIP/ OK Via: SIP/2.0/TLS :51682;branch=z9hG4bK1cd529ba From: <sip: @ >;tag=00260bd9669e07177ee0d51d-14f56f89 To: <sip:9@ >;tag= Date: Mon, 29 Mar :36:34 GMT Call-ID: 7747f400-bb01baf b6a12ac@ CSeq: 1001 NOTIFY Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 136
137 Case Study 3: No One Answers the Phone User Dials Remaining Digits 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 137
138 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) CUBE ( ) 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 138
139 Case Study 3: No One Answers the Phone CUCM Sends an INVITE to the CUBE 03/29/ :36: //SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to :[5060]: INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1515b From: "Test User 1" To: Date: Mon, 29 Mar :36:41 GMT Call-ID: 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: :5060>;method="notify;event=telephone-event;duration=500" Cisco-Guid: Session-Expires: 1800 P-Asserted-Identity: "Test User 1" Contact: Max-Forwards: 69 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 139
140 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) CUBE ( ) 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 200 OK (SUBSCRIBE) INVITE 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 140
141 Case Study 3: No One Answers the Phone CUBE Replies With a 183 Session Progress W/ SDP 03/29/ :36: //SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 1568 from :[5060]: SIP/ Session Progress Via: SIP/2.0/UDP :5060;branch=z9hG4bK1515b From: "Test User 1" <sip: @ >;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd To: <sip: @ >;tag=de1eff8-0 Date: Mon, 29 Mar :37:23 GMT Call-ID: 7c0ca800-bb01baf9-1468e-3b6a12ac@ CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: <sip: @ >;party=called;screen=no;privacy=off Contact: <sip: @ :5060> Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: multipart/mixed;boundary=uniqueboundary Mime-Version: 1.0 Content-Length: uniqueboundary 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 141
142 Case Study 3: No One Answers the Phone CUBE Replies With a 183 Session Progress W/ SDP Content-Type: application/sdp Content-Disposition: session;handling=required v=0 o=ciscosystemssip-gw-useragent IN IP s=sip Call c=in IP t=0 0 m=audio RTP/AVP c=in IP 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/8000 a=fmtp: a=rtpmap:101 telephone-event/8000 a=fmtp: uniqueboundary Content-Type: application/x-q931 Content-Disposition: signal;handling=optional Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 142
143 Case Study 3: No One Answers the Phone Unified CM Sends a 180 Ringing to the IP Phone 03/29/ :36: //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to on port index 2321 SIP/ Ringing Via: SIP/2.0/TLS :51682;branch=z9hG4bK1636ab61 From: "Test User 1" <sip: @ >;tag=00260bd9669e07147bcb3aac-3cda8f0c To: <sip:9@ ;user=phone>;tag=97903bc0-a3de-4a15-ba27-44c81fe3adcd Date: Mon, 29 Mar :36:33 GMT Call-ID: 00260bd9-669e000b-588c0c2b-2193e2a3@ CSeq: 101 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence Contact: <sip:9@ :5061;transport=tls> Call-Info: <urn:x-cisco-remotecc:callinfo>; security= NotAuthenticated; orientation= to; ui-state= ringout; gci= ; callinstance= 1 Send-Info: conference Remote-Party-ID: <sip: @ >;party=called;screen=no;privacy=off Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 143
144 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) CUBE ( ) 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 200 OK (SUBSCRIBE) 180 Ringing INVITE 100 Trying 183 Session Progress 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 144
145 Case Study 3: No One Answers the Phone Phone Keeps Ringing Timestamps Jump from 10:36:42 to 10:37:32 No SIP Signaling for 50 seconds 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 145
146 Case Study 3: No One Answers the Phone Phone Sends a CANCEL 03/29/ :37: //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from on port index 2321 with 422 bytes: CANCEL sip:9@ ;user=phone SIP/2.0 Via: SIP/2.0/TLS :51682;branch=z9hG4bK1636ab61 From: "Test User 1" <sip: @ >;tag=00260bd9669e07147bcb3aac-3cda8f0c To: <sip:9@ ;user=phone> Call-ID: 00260bd9-669e000b-588c0c2b-2193e2a3@ Max-Forwards: 70 Date: Mon, 29 Mar :37:32 GMT CSeq: 101 CANCEL User-Agent: Cisco-CP9951/9.0.1 Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 146
147 Case Study 3: No One Answers the Phone Unified CM Sends a 200 OK for the CANCEL 03/29/ :37: //SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to on port index 2321 SIP/ OK Via: SIP/2.0/TLS :51682;branch=z9hG4bK1636ab61 From: "Test User 1" <sip: @ >;tag=00260bd9669e07147bcb3aac-3cda8f0c To: <sip:9@ ;user=phone> Date: Mon, 29 Mar :37:32 GMT Call-ID: 00260bd9-669e000b-588c0c2b-2193e2a3@ CSeq: 101 CANCEL Content-Length: Cisco and/or its affiliates. All rights reserved. Cisco Public 147
148 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) CUBE ( ) NOTIFY 200 OK (NOTIFY) CANCEL 200 OK (CANCEL) 487 Request Cancelled ACK CANCEL 200 OK (CANCEL) 487 Request Cancelled ACK 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 148
149 Case Study 3: No One Answers the Phone Phone 1 Unified CM SBC (CUBE) INVITE (w/ OFFER) INVITE (no SDP) CUBE INVITE w/ OFFER SBC SP SBC SIP SP 180 Ringing (no SDP) 183 Session Progress (w/ OFFER) 183 Session Progress (w/ ANSWER) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 149
150 Case Study 3: No One Answers the Phone Phone 1 Unified CM SBC (CUBE) INVITE (w/ OFFER) INVITE (no SDP) CUBE INVITE w/ OFFER SBC SP SBC SIP SP 183 Session Progress (w/ OFFER) 183 Session Progress (w/ ANSWER)??? (w/ ANSWER)??? (w/ ANSWER) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 150
151 Case Study 3: No One Answers the Phone Phone 1 Unified CM SBC (CUBE) INVITE (w/ OFFER) INVITE (no SDP) CUBE INVITE w/ OFFER SBC SP SBC SIP SP 183 Session Progress (w/ OFFER) 183 Session Progress (w/ ANSWER) 183 Session Progress (w/ ANSWER) PRACK (w/ ANSWER) 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 151
152 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 cube(conf-serv-sip)#rel1xx? disable Disables reliable-provisional responses require Requires reliable-provisional responses supported Supports reliable-provisional responses 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 152
153 Case Study 3: No One Answers the Phone Unified CM SIP Profile Configuration 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 153
154 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) CUBE ( ) 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 200 OK (SUBSCRIBE) 183 Session Progress INVITE 100 Trying 183 Session Progress PRACK 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 154
155 Case Study 4: Calls to S4B Clients Fail When a user dials a S4B client from a video-enabled Cisco 9951 phone, the call fails. Call is from to Cisco and/or its affiliates. All rights reserved. Cisco Public 155
156 Case Study 4: Live Demo
157 Case Study 4: Calls to S4B 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 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 157
158 Case Study 4: Calls to S4B Clients Fail For more information: Visit to download the SIP Normalization and Transparency Developer Guide 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public 158
159 Case Study 5: Live Demo
160 Cisco Spark How Questions? Use Cisco Spark to communicate with the speaker after the session 1. Find this session in the Cisco Live Mobile App 2. Click Join the Discussion 3. Install Spark or go directly to the space 4. Enter messages/questions in the space cs.co/ciscolivebot# 2018 Cisco and/or its affiliates. All rights reserved. Cisco Public
161 Please complete your Online Session Evaluations after each session Complete 4 Session Evaluations & the Overall Conference Evaluation (available from Thursday) to receive your Cisco Live T-shirt All surveys can be completed via the Cisco Live Mobile App or the Communication Stations Complete Your Online Session Evaluation Don t forget: Cisco Live sessions will be available for viewing on-demand after the event at Cisco and/or its affiliates. All rights reserved. Cisco Public
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