Troubleshooting SIP with Cisco Unified Communications

Transcription

1 Troubleshooting SIP with Cisco Unified Communications BRKUCC-2932 Follow us on Twitter for real time updates of the #CLEUR

2 Housekeeping We value your feedback- don't forget to complete your online session evaluations after each session & the Overall Conference Evaluation which will be available online from Thursday Visit the World of Solutions and Meet the Engineer Visit the Cisco Store to purchase your recommended readings Please switch off your mobile phones After the event don t forget to visit Cisco Live Virtual: Follow us on Twitter for real time updates of the #CLEUR 2

3 Agenda Introduction Session Initiation Protocol (SIP) Overview Troubleshooting Tools Unified CM Tracing Cisco Unified Border Element (CUBE) Tracing Sample Call Flows / Case Studies 4

4 SIP Protocol Overview

5 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 6

6 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 7

7 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) 8

8 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 9

9 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 10

10 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-CUCM8.6 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER 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: 864 Content-Type: application/sdp 11

11 SIP INVITE Method 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-CUCM8.6 Allow: INVITE, SIP OPTIONS, Method INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER 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: 864 Content-Type: application/sdp 12

12 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-CUCM8.6 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER 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: 864 Content-Type: application/sdp 13

13 SIP Responses Response Code 1xx 2xx 3xx 4xx 5xx 6xx Description 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 Server Error Server failed to fulfill an apparently valid request Global Failure Request is invalid at any server Example 100 Trying 180 Ringing 183 Session Progress 200 OK 202 Acceptable 300 Multiple Choices 301 Moved Permanently 302 Moved Temporarily 401 Unauthorized 406 Not Acceptable 486 Busy Here 487 Request Terminated 488 Not Acceptable Here 503 Service Unavailable 600 Busy Everywhere 603 Decline 14

14 Basic SIP Call Setup Phone 1 Unified CM INVITE 200 OK ACK Session Established BYE 200 OK 15

15 Basic SIP Call Setup with B2BUA (Unified CM) Phone 1 Unified CM SBC Phone (CUBE) 2 INVITE INVITE CUBE 200 OK ACK Session Established 200 OK ACK BYE 200 OK BYE 200 OK 16

16 Basic SIP Call Setup with Unified CM and CUBE Phone 1 Unified CM SBC (CUBE) SP SBC SIP SP INVITE INVITE CUBE INVITE SBC 200 OK ACK 200 OK ACK 200 OK ACK Session Established BYE BYE BYE 200 OK 200 OK 200 OK 17

17 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 18

18 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. 19

19 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 20

20 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 97 21

21 Media Negotiation 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. For Delayed Offer, the answer is typically sent in the ACK. 22

22 Early Offer Phone 1 Unified CM INVITE with SDP - Offer 200 OK with SDP - Answer ACK (no SDP) Session Established BYE 200 OK 23

23 Delayed Offer Phone 1 Unified CM INVITE (no SDP) 200 OK with SDP - Offer ACK with SDP - Answer Session Established BYE 200 OK 24

24 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) 25

25 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 26

26 Media Re-negotiation Re-INVITE Either UA involved in a call can re-invite an existing dialog to renegotiate 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. 27

27 Media Re-negotiation Re-INVITE INVITE SIP/2.0 Via: SIP/2.0/TLS :5061;branch=z9hG4bK901f9c72c19221 From: "Paul Giralt" b82e2c213ca To: Date: Wed, 11 Jan :08:51 GMT Call-ID: Supported: 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: 104 INVITE Max-Forwards: 70 Expires: 180 Allow-Events: presence Call-Info: <urn:x-cisco-remotecc:callinfo>; security= Authenticated; orientation= from; gci= ; callinstance= 2 Remote-Party-ID: "Paul Giralt" <sip: @ >;party=calling;screen=yes;privacy=off Contact: <sip: @ :5061;transport=tls> Content-Type: application/sdp Content-Length:

28 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:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx XXXXXXXXX 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:

29 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" b82e2c213ca To: Date: Wed, 11 Jan :08:52 GMT Call-ID: Supported: 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: 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: 0 30

30 Media Re-negotiation Re-INVITE Offer in 200 OK SIP/ OK Via: SIP/2.0/TLS :5061;branch=z9hG4bK901fac34c0fb1b From: "Paul Giralt" b82e2c213ca To: Call-ID: Date: Wed, 11 Jan :08:52 GMT CSeq: 106 INVITE Server: Cisco-CPCIUS/9.2.1 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-service-control,x-cisco-srtp-fallback,x-cisco-monrec,x-cisco-config,x-cisco-sis ,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 31

31 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 32

32 Media Re-negotiation Re-INVITE Answer in ACK ACK SIP/2.0 Via: SIP/2.0/TLS :5061;branch=z9hG4bK901fb064465a06 From: "Paul Giralt" b82e2c213ca 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:

33 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:xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx XXXXXXXXX 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:

34 Troubleshooting Tools

35 SIP Troubleshooting Tools Unified CM / SME Tools: - Real Time Monitoring Tool / Session Trace - TranslatorX Wireshark 36

36 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 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 Can only be used on calls for which traces still exist on the server (Unified CM 9.0 will allow viewing traces that have been archived off-server) 37

37 RTMT Session Trace Tool 38

38 RTMT Session Trace Tool Call Flow Diagram 39

39 RTMT Session Trace Tool Hover to get information about message Click See SIP message to see actual SIP message 40

40 RTMT Session Trace Tool SIP Message Details 41

41 TranslatorX Tool Features Parses through Unified CM CCM/SDI Trace Files Decodes SIP, SCCP, H.323, MGCP, 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: NOTE: Do not call TAC for support on TranslatorX 42

42 TranslatorX Tool 43

43 TranslatorX Tool Call List Window 44

44 TranslatorX Tool Call List Filtering Select Each Call and click Generate Filter button 45

45 TranslatorX Tool CDR View 46

46 TranslatorX Tool CDR View 47

47 TranslatorX Tool Message Filters 48

48 TranslatorX Tool 49

49 TranslatorX Tool 50

50 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 51

51 Wireshark 52

52 Wireshark VoIP Call Analysis 53

53 Wireshark VoIP Call Ladder Diagram 54

54 Wireshark How to gather a trace? Both Unified CM and IOS provide a mechanism to gather a packet capture Will be covered later 55

55 Unified CM Tracing Configuration

56 Unified CM Trace Configuration SIP messaging in Unified CM is written to the CCM/SDI trace file when appropriate trace levels are set Configured from Cisco Unified Serviceability > Trace > Configuration or by using AnalysisManager Unified CM 9.0 will combine SDI and SDL traces into the SDL traces, so be on the lookout. 57

57 Unified CM Trace Configuration Select the Server Select Service Group Select the Service on Which Trace Needs to Be Enabled 58

58 Unified CM Trace Configuration 1. Press Set Default Updates All Servers in This Cluster with These Settings 2. Set to Detailed 59

59 Unified CM Trace Configuration Enable SIP Stack Trace is NOT needed to see SIP Messages. Do not enable SIP Stack Trace unless directed by TAC 60

60 Unified CM Trace Configuration Can Also Use the Troubleshooting Trace Settings Page in CallManager Serviceability (Trace > Troubleshooting Trace Settings) 61

61 Trace Collection Various Ways to Collect Trace Files RTMT Analysis Manager RTMT Remote Browse RTMT Collect Files RTMT Query Wizard OS CLI (file get or file tail) 62

62 Gathering a Packet Capture from Unified CM Use the Platform CLI command utils network capture a d m i n : utils n e t w o r k c a p t u r e? S y n t a x : u t i l s n e t w o r k c a p t u r e [ o p t i o n s ] o p t i o n s o p t i o n a l p a g e, n u m e r i c, f i l e f n a m e, c o u n t num, s i z e b y t e s, src addr, d e s t addr, p o r t num, host p r o t o c o l addr a d m i n : utils n e t w o r k c a p t u r e f i l e c a p t u r e f i l e c o u n t s i z e A L L h o s t ip E x e c u t i n g c o m m a n d w i t h o p t i o n s : s i z e = A L L c o u n t = i n t e r f a c e = e t h 0 s r c = d e s t = p o r t = ip= a d m i n : file list a c t i v e l o g p l a t f o r m / c l i c a p t u r e f i l e. c a p dir c o u n t = 0, f i l e c o u n t = 1 a d m i n : file get a c t i v e l o g p l a t f o r m / c l i / c a p t u r e f i l e. c a p P l e a s e w a i t w h i l e t h e s y s t e m i s g a t h e r i n g f i l e s i n f o... d o n e. Sub-directories w e r e n o t t r a v e r s e d. N u m b e r o f f i l e s a f f e c t e d : 1 T o t a l s i z e i n B y t e s : 2 4 T o t a l s i z e i n K b y t e s : W o u l d y o u l i k e t o p r o c e e d [ y / n ]? y 63

63 Cisco Unified Border Element (CUBE) Tracing Configuration

64 CUBE Debugging CUBE / IOS Tools: - IOS debugs - IOS show commands - Per-call trace - Packet export 65

65 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

66 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 67

67 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-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: :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: 0 68

68 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) 69

69 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: 2 70

70 CUBE Per-Call Debugging (PCD) Useful for CUBE under high call volume Available on all CUBE(Ent) ASR releases and in 15.1(2)T and later on ISR All the debug pertaining to a particular call goes into a buffer Trigger-points looks for specific info in the buffers to export the debug info to an output destination Can trigger based on user-defined criteria or log every call - SIP 4XX, 5XX, or 6XX Response - Q.850 Cause code - Call Admission Control limits 71

71 Reference CUBE Per-Call Debugging (PCD) PCD Configuration 1. Define buffers and buffer sizes per-call num-buffer <num> per-call buffer-size debug <num> 2. Turn per-call debugging on/off per-call shutdown per-call active debug per-call inactive 4. Export debug buffer content per-call export primary [flash ftp http pram rcp tftp] secondary [flash ftp http pram rcp tftp] 5. Show buffer content status show per-call stat show per-call buffer list 3. Set trigger points per-call trigger cause 1 per-call trigger cause 41 per-call trigger sip-message 404 per-call trigger sip-message Show buffer contents on console router#show per-call buffer content? < > Specify the buffer num router#show per-call buffer content 1 72

72 CUBE IP Traffic Capture Export Packet Data in PCAP format IP Traffic Export feature allows export of packets on an interface Configuration: 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 Usage: traffic-export interface g0/0 start traffic-export interface g0/0 stop traffic-export interface g0/0 copy scp:// /capture.pcap 73

73 Case Studies

74 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. 75

75 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 76

76 Case Study 1: Unable to place a call Use RTMT Session Trace Double-click to see message diagram Clearly shows the far-end sends back a 404 Not Found 77

77 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-CUCM8.6 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: 0 78

78 Case Study 1: Unable to place a call Troubleshoot call on CUBE 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 Check to see if the number matches a valid dial peer CUBE#show dialplan number Macro Exp.: No match, result=-1 79

79 Case Study 2: Unable to place a call #2 Problem Description A user reports that every time they call (919) , they get reorder (fast busy) tone. 80

80 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 two calls 81

81 Case Study 2: Unable to place a call #2 Use RTMT Session Trace Trace shows signaling from phone as well as to CUBE CUBE is responding with a 403 Forbidden 82

82 Case Study 2: Unable to place a call #2 Problem Description As of IOS 15.1(2)T, IOS will reject calls from unknown sources by default Can either disable the feature or add the list of permitted addresses voice service voip no ip address trusted authenticate allow-connections sip to sip sip voice service voip ip address trusted list ipv allow-connections sip to sip sip 83

83 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

84 Case Study 3: No One Answers the Phone Collect Traces Problem is reproducible, so generate a test call and then collect traces. 85

85 Case Study 3: No One Answers the Phone Use TranslatorX Problem is reproducible, so generate a test call and then collect traces. Select File > Open Folder 86

86 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces Search for called party number 87

87 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) 88

88 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: 0 89

89 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 90

90 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 91

91 Case Study 3: No One Answers the Phone Use TranslatorX to Analyze Traces 92

92 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-ciscoescapecodes,x-cisco-service-control,x-cisco-srtp-fallback,x-cisco-monrec,x-cisco-config,x-cisco-sis-5.0.0,xcisco-xsi Allow-Events: kpml,dialog Content-Length: 632 Content-Type: application/sdp Content-Disposition: session;handling=optional 93

93 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 94

94 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: 0 95

95 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:

96 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> 97

97 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@ ; fromtag=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> 98

98 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: 0 99

99 Case Study 3: No One Answers the Phone Unified CM Sends a SUBSCRIBE for KPML 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: 0 100

100 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) SIP Gateway ( ) INVITE 100 Trying REFER SUBSCRIBE 200 OK (REFER) 200 OK (SUBSCRIBE) 101

101 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"/> 102

102 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: 0 103

103 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 index 2321 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:

104 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> 105

105 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 index 2321 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:

106 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: 0 107

107 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) SIP Gateway ( ) INVITE 100 Trying REFER SUBSCRIBE 200 OK (REFER) 200 OK (SUBSCRIBE) NOTIFY 200 OK (NOTIFY) REFER 200 OK (REFER) 108

108 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"/> 109

109 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: 0 110

110 Case Study 3: No One Answers the Phone User Dials Remaining Digits 111

111 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) SIP Gateway ( ) 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 112

112 Case Study 3: No One Answers the Phone Digit Analysis Match 10:36: Digit analysis: match(pi="2", fqcn=" ", cn=" ",plv="5", pss="1stline:rtp_abbrdial:cisco:us Local:US RTP Local:US Long Distance:US International:VMPilotPartition", TodFilteredPss="1stLine:RTP_AbbrDial:Cisco:US Local:US RTP Local:US Long Distance:US International:VMPilotPartition", dd=" ",dac="1 ) 10:36: Digit analysis: analysis results 10:36: PretransformCallingPartyNumber= CallingPartyNumber= DialingPartition=GDP_GlobalE164_PSTN DialingPattern=\+1.[2-9]XX[2-9]XXXXXX FullyQualifiedCalledPartyNumber= DialingPatternRegularExpression=(+1)([2-9][0-9][0-9][2-9][0-9][0-9][0-9][0-9][0-9][0-9]) DialingWhere= PatternType=Enterprise PotentialMatches=NoPotentialMatchesExist DialingSdlProcessId=(0,0,0) PretransformDigitString= PretransformTagsList=ACCESS-CODE:SUBSCRIBER PretransformPositionalMatchList=+1: CollectedDigits= UnconsumedDigits= TagsList=ACCESS-CODE:SUBSCRIBER PositionalMatchList=+1: Voic box= Voic CallingSearchSpace=1stLine:RTP_AbbrDial 113

113 Case Study 3: No One Answers the Phone Digit Analysis Match Voic PilotNumber= RouteBlockFlag=RouteThisPattern RouteBlockCause=0 AlertingName= UnicodeDisplayName= DisplayNameLocale=1 OverlapSendingFlagEnabled=0 WithTags= WithValues= CallingPartyNumberPi=NotSelected ConnectedPartyNumberPi=NotSelected CallingPartyNamePi=NotSelected ConnectedPartyNamePi=NotSelected CallManagerDeviceType=NoDeviceType PatternPrecedenceLevel=Routine CallableEndPointName=[ d07dd6fdef] PatternNodeId=[9badd465-d20a-5bc edee47e8caf] AARNeighborhood=[] AARDestinationMask=[] AARKeepCallHistory=true AARVoic Enabled=false NetworkLocation=OffNet 114

114 Case Study 3: No One Answers the Phone Digit Analysis Match Calling Party Number Type=Cisco Unified CallManager Calling Party Numbering Plan=Cisco Unified CallManager Called Party Number Type=Cisco Unified CallManager Called Party Numbering Plan=Cisco Unified CallManager ProvideOutsideDialtone=false AllowDeviceOverride=false AlternateMatches= { Partition=US Long Distance { < Pattern=9.1[2-9]XX[2-9]XXXXXX PatternType=Translation TranslationPartition=[a6bd708e-ac4d-ae b90b987e5ad9] CallManagerDeviceType=NoDeviceType PatternPrecedenceLevel=PlDefault PatternRouteClass=RouteClassDefault RouteNextHopByCgpn=false > } } 115

115 Case Study 3: No One Answers the Phone Digit Analysis Match TranslationPatternDetails= PretransformCallingPartyNumber= CallingPartyNumber= DialingPartition=US Local DialingPattern=9.1877[2-9]XXXXXX FullyQualifiedCalledPartyNumber= DialingPatternRegularExpression=(9)(1877[2-9][0-9][0-9][0-9][0-9][0-9][0-9]) DialingWhere= PatternType=Translation PotentialMatches=NoPotentialMatchesExist DialingSdlProcessId=(0,0,0) PretransformDigitString= PretransformTagsList=ACCESS-CODE:SUBSCRIBER PretransformPositionalMatchList=9: CollectedDigits= UnconsumedDigits= TagsList=SUBSCRIBER PositionalMatchList= Voic box= Voic CallingSearchSpace= Voic PilotNumber= RouteBlockFlag=RouteThisPattern RouteBlockCause=1 AlertingName= 116

116 Case Study 3: No One Answers the Phone Digit Analysis Match UnicodeDisplayName= DisplayNameLocale=1 OverlapSendingFlagEnabled=0 WithTags= WithValues= CallingPartyNumberPi=NotSelected ConnectedPartyNumberPi=NotSelected CallingPartyNamePi=NotSelected ConnectedPartyNamePi=NotSelected CallManagerDeviceType=NoDeviceType PatternPrecedenceLevel=Routine CallableEndPointName=[bb6f140a-5fd4-179a-2cad-2a1d5eacca7e] PatternNodeId=[bb6f140a-5fd4-179a-2cad-2a1d5eacca7e] AARNeighborhood=[] AARDestinationMask=[] AARKeepCallHistory=true AARVoic Enabled=false NetworkLocation=OnNet Calling Party Number Type=Cisco Unified CallManager Calling Party Numbering Plan=Cisco Unified CallManager Called Party Number Type=Cisco Unified CallManager Called Party Numbering Plan=Cisco Unified CallManager ProvideOutsideDialtone=true AllowDeviceOverride=false AlternateMatches= 117

117 Case Study 3: No One Answers the Phone CUCM Sends an INVITE to the PSTN Gateway 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: 0 118

118 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) SIP Gateway ( ) 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 119

119 Case Study 3: No One Answers the Phone Gateway 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 120

120 Case Study 3: No One Answers the Phone Gateway 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:

121 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= ; call-instance= 1 Send-Info: conference Remote-Party-ID: <sip: @ >;party=called;screen=no;privacy=off Content-Length: 0 122

122 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) SIP Gateway ( ) 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 123

123 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 124

124 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: 0 125

125 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: 0 126

126 Case Study 3: No One Answers the Phone IP Phone ( ) Unified CM ( ) SIP Gateway ( ) NOTIFY 200 OK (NOTIFY) CANCEL 200 OK (CANCEL) 487 Request Cancelled ACK CANCEL 200 OK (CANCEL) 487 Request Cancelled ACK 127

127 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 SEP 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 128

128 Recommended Reading Please visit the Cisco Store for suitable reading.

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