Fixing SIP Problems with UC Manager's SIP Normalization Tools
|
|
- Miles Jenkins
- 6 years ago
- Views:
Transcription
1
2 Fixing SIP Problems with UC Manager's SIP Normalization Tools Mark Stover
3 Why have this session? More systems than ever use SIP I counted 103 SIP Products on SIP Wikipedia Page Google Search for SIP Server yields 2.8 Million Hits Many of the SIP bits don t quite match up Hence, the need for Interoperability events Things in the real world don t go the way of data sheets and Interoperability Forums! 3
4 What this Session is About Fact 1: SIP is a Standard Fact 2: SIP Configurations are not standardized Which headers are included Format of data in headers (URIs, etc.) Ordering of header fields Contents of the SIP Message Body What do we do when Fact #1 and Fact #2 are at odds in our deployment? We have a tool in our toolbox: SIP Transparency and Normalization 4
5 Typical Interop Scenario 5
6 Agenda Brief review of SIP When things don t work Overview of SIP Transparency and Normalization Overview of Lua Normalization Scripts Case Study Some Common Scripting Forms Case Study 2 Conclusion 6
7 Brief Review of SIP
8 Basic Design SIP is a Client-Server Protocol Clients send requests, receive responses Servers receive requests, send responses Modeled after HTTP Text Encoded Protocol Client request Server Each request invokes method on server Main purpose of request response Messages contain bodies 8
9 SIP Methods and Messages Call signaling performed by SIP Methods Six Original SIP Methods: INVITE ACK OPTIONS BYE CANCEL REGISTER SIP Messages have distinct parts: IP/TCP/UDP Envelope SIP Header SIP Message Body MIME-Encoded Session Description Protocol (SDP) May contain other data 9
10 SIP Methods INVITE Invites a participant to a session idempotent - reinvites for session modification BYE Ends a client s participation in a session CANCEL Terminates a search For Your Reference OPTIONS Queries a participant about their media capabilities, and finds them, but doesn t invite PING identifies reachability ACK For reliability and call acceptance REGISTER Informs a SIP server about the location of a user 10
11 SIP Message Syntax Many header fields from http Payload contains a media description SDP Session Description Protocol INVITE sip:alice@company.com SIP/2.0 From: Bob <sip:bob@university.edu> To: Alice <sip:alice@company.com> Via: SIP/2.0/UDP pc.university.edu Call-ID: @ Content-type: application/sdp CSeq: 4711 INVITE Content-Length: 187 v=0 o=ccm-sip IN IP s=sip Call c=in IP m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=ptime:20 a=mid:1 c=in IP6 2001:0db8:aaaa::0987:65ff:fe01:234b m=audio RTP/AVP 0 a=mid:2 11
12 Negotiating the Session For Your Reference Called party receives SDP offered by caller Each stream can be accepted rejected Accepting involves generating an SDP listing same stream port number and address of called party subset of codecs from SDP in request Rejecting indicated by setting port to zero Resulting SDP returned in 200 OK Audio stream accepted, PCMU only Video stream rejected, Port 0 v=0 o=user IN IP t=0 0 m=audio 3456 RTP/AVP 0 c=in IP m=video 0 RTP/AVP 86 c=in IP Media can now be exchanged 12
13 SIP Responses Look much like requests Headers, bodies Differ in top line Status Code Numeric, Meant for computer processing Protocol behavior based on 100s digit Other digits give extra info Reason Phrase Text phrase for humans Can be anything Status Code Classes (1XX): Informational (2XX): Success (3XX): Redirection (4XX): Client Error (5XX): Server Error (6XX): Global Failure Two groups : Provisional (Not reliable) : Final, Definitive Example 200 OK 180 Ringing 13
14 SIP Transactions Fundamental unit of messaging exchange Request Zero or more provisional responses Usually one final response Maybe ACK All signaling composed of independent transactions Transactions identified by Cseq Sequence number Method tag 14
15 When things don t work
16 Identifying A Problem Goal is to make two SIP systems talk Both systems already configured with: Appropriate Network Configurations Trunk configuration to reach the other system Routing (dial plan) information in place Make a call From: 1001 on System A; To: 2001 on System Z Both phones exist, are configured, and are able to make other calls And wait 16
17 Symptoms Possible failure modes: Wait forever and get fast busy Call rejected right away Calls work, but other services (e.g. MWI) fail These can be symptoms of numerous problems No bandwidth No DNS Bad Codecs SIP Header Mismatches Etc. Unfortunately, it will take some troubleshooting to isolate the issue 17
18 Gather Information Logs are good: Will help you determine if SIP is the problem May not reflect what is really on the wire May not include the header level detail Packet Capture is your friend Various ways to gather traces Further discussion just ahead Review Paul Giralt s SIP Troubleshooting session for many more details: BRKUCC-2932 Troubleshooting SIP with Cisco Unified Communications It is scheduled for Thursday Afternoon 18
19 Getting SIP Messages Three main sources of SIP Message Information 1. Unified CM Trace Files 2. Unified CM Network Capture utils network capture 3. Network Packet Capture (Wireshark) 19
20 Example of Unified CM Trace File 17:38: //SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from on port with 1872 bytes: INVITE SIP/2.0 Via: SIP/2.0/TCP :5060;branch=z9hG4bK From: "Alice" To: Date: Wed, 12 Oct :38:59 GMT Call-ID: Supported: 100rel,timer,resource-priority,replaces,sdp-anat Cisco-Guid: User-Agent: Cisco-SIPGateway/IOS-12.x CSeq: 101 INVITE Timestamp: Expires: 180 Allow-Events: telephone-event Content-Type: multipart/mixed;boundary=uniqueboundary Mime-Version: 1.0 Content-Length: uniqueboundary 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/
21 Using Unified CM Network Capture admin:utils network capture size 1500 port 5060 file testsipcap verbose Executing command with options: size=1500 count=1000 interface=eth0 src= dest= port=5060 ip= admin:file list activelog platform/cli/ testsipcap.cap dir count = 0, file count = 1 admin:file get activelog platform/cli/testsipcap.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: 6040 Total size in Kbytes: Would you like to proceed [y/n]? y SFTP server IP: SFTP server port [22]: User ID: admin Password: ******** Download directory: Downloads. Transfer completed. 21
22 Using Wireshark 22
23 Determine Needed Results 1. Make calls in both directions: Get SIP Captures of test calls in both directions 2. Traces may give you a clue: Mismatch in domain names No domain in one direction Mailbox you want is last redirect instead of first in list 3. May have to research each system s SIP trunk requirements Compare it to the other vendor s normal operation 4. Use your research and troubleshooting to determine the fix: Change the domain name of messages from incorrectly configured system Add a missing domain Remove headers that cause a failure 23
24 Write, Test, and Deploy Use the desired result to formulate a plan Create Normalization Script that process appropriate SIP headers Test against traffic on a SIP trunk that does not carry production traffic Deploy to production trunk and verify 24
25 Overview of SIP Transparency & Normalization
26 Goals of SIP Transparency & Normalization Provide an interface for customization of SIP messages Initially conceived for Cisco Unified CM Session Management Edition (SME) Also supports Cisco Unified Communications Manager without SME Available in Release 8.5 and later Include: A Lua execution environment SIP Transparency & Normalization APIs Support: Transparent passing of SIP information from one call leg to another Normalizing SIP Messages to provide interoperability 26
27 SIP Transparency Unified CM is a Back to Back User Agent (B2BUA) In a Session Management role, Unified CM will (by default ) insert itself in the call Will become the call agent for next leg Will remove any unsupported headers on the next leg Transparency allows SIP information to be passed from one call leg to another What that means? Transparency allows 3rd-Party Headers to pass through Unified CM 27
28 SIP Normalization The process of transforming inbound and outbound SIP messages Inbound normalization makes the SIP message useable by Unified CM For Example, how can we handle SIP redirecting numbers? Unified CM uses the Diversion header for redirecting number(s) Other SIP devices use the History-Info header for this purpose Normalization can transform History-Info headers into Diversion headers Outbound normalization makes the SIP message useable by another SIP device Use normalization to transform Unified CM s Diversion headers into History-Info headers for another SIP PBX 28
29 Normalization Script Examples Reorder codecs in the SDP of an early offer Remove specific headers such as Cisco-Guid Mask the number to E.164 in a Diversion header to meet Service Provider requirements Fix the domain name of a system with the wrong one Convert IP addresses to domain names Add content to the SIP Message Body 29
30 What can a normalization script change? Manipulate almost every aspect of a SIP message Currently, SIP Normalization can change: The request URI The response code and phrase SIP headers SIP parameters Content bodies SDP 30
31 What can a transparency script do? To provide transparency, the script has to pass SIP information Almost any information in a SIP message can be passed through Currently, SIP Transparency can manage: SIP headers SIP parameters Content bodies 31
32 Case Study-Problem Statement Customer has several PBXs trunked to a Unified CM cluster Unified CM SIP trunked to a 3rd-party voice mail system After multiple call forwards, some calls are sent to voice mail Most calls go to the correct voice mail box Calls from one PBX did not In the broken case: Reaching the greeting for the station that finally forwarded the call to voice mail Not reaching the voice mail of the station originally called Will solve this problem with SIP Normalization in just a little while 32
33 Overview of Lua
34 What is Lua? A powerful, fast, lightweight, embeddable scripting language A fast language engine with a small footprint that can embed easily into other applications Lua has a simple and well documented API that allows strong integration with code in other languages Adding Lua to an application does not bloat it Many more details about Lua can be found online:
35 A Brief Lua Tutorial This is not a programming course! Will cover some Lua basics to allow writing SIP T&N Scripts Will briefly consider: Lua Data Types Lua Tables Lua Control Structures Unified CM Support for Lua 35
36 Lua Data Types Lua has the typical data types you would expect: Numbers Strings Boolean (true or false) Lua has one data type you might not have heard of: Tables Tables are the only aggregate data type available in Lua 36
37 Lua Tables Tables are used for storing collections lists, arrays, and associative arrays Tables can contain other objects including numbers, strings, or tables Tables created using a pair of curly brackets { } t = { 1,1,2,3,5,8,13 } t[1] == 1 Note that table indexes begin at 1 Methods (functions) exist to insert and remove table elements Library functions allow iterating over the contents of a table 37
38 Using Lua Tables for SIP Headers Tables are a key part of how Lua can process SIP headers Tables are useful when more than one of a specific header is present For example: History-Info: <sip:userb@hostb?reason=sip;cause=408>;index=1 History-Info: <sip:userc@hostc?reason=sip;cause=302>;index=1.1 History-Info: <sip:userd@hostd>;index=1.1.1 Values from all three headers can be stored in a Table (history_info) history_info[1] == "<sip:userb@hostb?reason=sip;cause=408>;index=1" history_info[2] == "<sip:userc@hostc?reason=sip;cause=302>;index=1.1" history_info[3] == "<sip:userd@hostd>;index=1.1.1" 38
39 Lua Control Structures Lua has the typical programmatic control structures There are four main forms: 1. While: conditional looping statement with the form: while <exp> do <block> end 2. Repeat: conditional looping statement with the form: repeat <block> until <exp> 3. If: selection statement with the form: if <exp> then <block> { elseif <exp> then <block> } [ else <block> ] end 4. For: iterating statement (see the next slide) 39
40 Looping With For The two forms that for can take The first is for numerical iteration for <var> = <from_exp>, <to_exp> [, <step_exp>] do <block> end for count = 1,3 do print(count) end The second is for sequential iteration for <var> {, <var>} in <explist> do <block> end Example: Use for to print the contents of a table: For is passed an function, pairs(), that supplies the values of each iteration for key,value in pairs({10, math.pi, "banana"}) do print(key, value) end banana 40
41 pairs and ipairs Part of the standard Lua library pairs() function iterates over all key-value pairs items are NOT returned in a defined order for key,value in pairs(t) do print(key,value) end pi banana yellow ipairs() function iterates over only index-value pairs Returned in numeric order of the indices; Non-integer keys are skipped for index,value in ipairs(t) do print(index,value) end
42 Handy Lua Bits Available on Unified CM tostring() is handy for getting numbers back to strings for SIP headers Comments can be single or multiple lines: -- This is a comment --[[ This is a comment that crosses multiple lines --]] 42
43 Lua Demo Using Lua on your computer Running the Lua environment Hello World 43
44 CUCM Lua Support Cisco SIP Lua Environment supports the following libraries: The complete string library A subset of the base library Other Lua libraries are not supported Cisco SIP Lua Environment provides Global environment for the scripts to use Default Lua global environment (_G) is not available to SIP T&N scripts Supported base library functions: ipairs pairs next unpack error type tostring 44
45 Pop Quiz! True or False: Cisco created Lua just for processing SIP messages? A. True B. False Which Lua Libraries does Unified CM Normalization Scripts support? A. None of Them B. All of Them C. Complete String Library D. Subset of Base Library E. Both C & D 45
46 Overview of Normalization Scripts
47 Putting SIP Normalization to Work Using message handlers to manipulate SIP messages Message Handler name tells you: When the Handler will be invoked What type of message the Handler is for You want your script to process INVITEs received by Unified CM: Script should have an inbound_invite message handler Corresponding message handler invoked anytime an inbound INVITE is received A single parameter called msg represents the SIP Message in script Scripts use Cisco SIP Message API library to access and manipulate the msg parameter Let s try looking at a diagram 47
48 How a Normalization Script Gets Run 48
49 How a Normalization Script Gets Run 49
50 How a Normalization Script Gets Run 50
51 How a Normalization Script Gets Run 51
52 How a Normalization Script Gets Run 52
53 How a Normalization Script Gets Run 53
54 How a Normalization Script Gets Run 54
55 How a Normalization Script Gets Run 55
56 Let s Start with a Simple Script Need to convert incoming History-Info headers into Diversion headers Script will run when Unified CM receives an INVITE Need to remove Cisco-Guid from outgoing headers Script will run when Unified CM sends an INVITE 56
57 Our First SIP Normalization Script M = {} function M.inbound_INVITE(msg) msg:converthitodiversion() end function M.outbound_INVITE(msg) msg:removeheader("cisco-guid") end return M 57
58 Focus on SIP Normalization Script - 1 M = {} Creates an empty Lua Table called M for all Handlers M is also the name of the Lua Module function M.inbound_INVITE(msg) msg:converthitodiversion() end function M.outbound_INVITE(msg) msg:removeheader("cisco-guid") end return M 58
59 Focus on SIP Normalization Script - 2 M = {} function M.inbound_INVITE(msg) msg:converthitodiversion() end Inbound INVITE Message Handler Inbound SIP Message accessed through msg Invokes an API call to perform the actual conversion function M.outbound_INVITE(msg) msg:removeheader("cisco-guid") end return M 59
60 Focus on SIP Normalization Script - 3 M = {} function M.inbound_INVITE(msg) msg:converthitodiversion() End function M.outbound_INVITE(msg) msg:removeheader("cisco-guid") end Outbound INVITE Message Handler Outbound SIP Message accessed through msg Invokes API call to remove a header (in this case, Cisco-Guid) return M 60
61 Focus on SIP Normalization Script - 4 M = {} function M.inbound_INVITE(msg) msg:converthitodiversion() end function M.outbound_INVITE(msg) msg:removeheader("cisco-guid") end return M Line is required Returns the Lua Table containing the message handlers to Unified CM execution environment Cisco SIP Lua Environment uses Table M to identify the message handlers 61
62 SIP Message Handler Formalities Each Transparency and Normalization script provides: Set of call-back functions to manipulate SIP messages Call-back functions are called message handlers The message handler s name indicates when a handler is invoked Only one Transparency AND Normalization Script per SIP Trunk Must define all message handlers in that single script Mix and match methods and directions in a single script Handlers for requests and responses have slightly different formats Will be covered next 62
63 Request Message Handlers Request message handler is named by combining: the SIP message direction AND the SIP method name Identify the method name from the 'request line' of the SIP message Request format: <direction>_<method> Examples: inbound_invite outbound_update 63
64 Response Message Handlers Response message handler is named by combining: the message direction PLUS the response code AND the SIP method Identify the method name from the CSeq header Response format: <direction>_<response code>_<method> Examples: inbound_183_invite inbound_200_invite outbound_200_update 64
65 Using Wild Cards in Message Handler Names For Request Messages A wildcard ANY can be used in place of <method> <direction> does not support a wild card For Response Messages: A wildcard ANY can be used in place of <method> A wildcard ANY can be used in place of <response code> <method> and <response code> can both be ANY <direction> does not support a wild card Cannot have a wildcard ANY <method> with a specific <response code> A wildcard character X can be used in <response code> 65
66 Examples of Wild Cards Valid Request Message Handler Names M.inbound_INVITE M.inbound_ANY M.outbound_ANY Valid Response Message Handler Names M.inbound_183_INVITE M.inbound_18X_INVITE M.outbound_ANY_INVITE M.outbound_ANY_ANY Invalid Response Names M.inbound_183_ANY 66
67 Rules for picking a message handler For Your Reference Unified CM uses these rules to choose a message handler: Message handlers are case-sensitive The direction is either inbound or outbound The direction is always written as lowercase The message direction is relative to Unified CM Note: The message direction has nothing to do with the dialog direction of the SIP session The method name in the SIP message is converted to uppercase to pick the message handler Longest match criteria: Unified CM uses the longest-match to choose the message handler A script has two message handlers: inbound_any_any and inbound_183_invite A 183 response is received by Unified CM The inbound_183_invite handler will be executed since it is the longest match 67
68 Built-In Normalization Scripts 68
69 APIs for SIP and SDP Normalization For Your Reference SIP Messages APIs: Allows script to manipulate the SIP message SDP APIs: Allows script to manipulate the SDP SIP Pass Through APIs: Allows script to pass information from one call leg to another SIP Utility APIs: Utilities to manipulate header data such a parsing URIs into a SIP URI object SIP URI APIs: Allows script to manipulate the parsed SIP URI object Trace APIs: Allows script to enable, disable and manage tracing Script Parameters API: Allows script to obtain trunk or line specific parameters 69
70 SIP Objects and Normalization Which SIP methods can be normalized? You can invoke a scripts based on any Method that Unified CM handles Which SIP headers can your script access? You can access any headers in the message that invokes the script Which lines in the message s SDP can you access? Your script can access any of the SDP lines For normalization, scripts can manipulate almost every aspect of a SIP message 70
71 SIP Objects and Transparency support Transparency is limited to INVITE dialogs on SIP trunks Transparency scripts can pass almost any information in a SIP message SIP Headers SIP Parameters Content Bodies These SIP objects do not support Transparency scripts SUBSCRIBE dialogs PUBLISH out-of-dialog REFER out-of-dialog unsolicited NOTIFY MESSAGE 71
72 Case Study
73 Case Study Calls going to the wrong mail box Customer has several PBXs trunked to Cisco Unified CM Unified CM interfaced to a 3rd-party voice mail system via SIP Calls sent to voice mail after multiple call forwards Most calls were going to the correct voice mail box Calls from one PBX were not In the broken case, calls were going to the voice mail box of the last station the call was forwarded to Let s look in detail at these call flows 73
74 Case study call flow 5 Call from PSTN for x Call from PSTN for x Call Forward All to x No Answer to Voice Mail 2 Call Forward All to x No Answer to Voice Mail Bad Result 8 Greeting for x Greeting for x2100 Good Result 74
75 Problem: SIP header from call to wrong mail box INVITE SIP/2.0 Via: SIP/2.0/TCP :5060;branch=z9hG4bK12b5cc229a69621 From: "PSTN" To: Date: Wed, 19 Dec :45:01 GMT Call-ID: Supported: 100rel,timer,resource-priority,replaces User-Agent: Cisco-CUCM8.5 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Call-Info: <sip: :5060>;method="notify;event=telephone-event;duration=500" Cisco-Guid: Session-Expires: 1800 Diversion: Diversion: Contact: Content-Length: 0 Call goes to x1200 greeting instead of x
76 What the script will have to accomplish Keep it simple & just remove the headers we don t need for voice mail 76
77 Minimal Normalization Script for outbound INVITEs M = {} function M.outbound_INVITE(msg) -- Process outbound INVITES to VM -- Process INVITE to normalize it... end return M 77
78 Add logic to remove extra Diversion Headers local DiversArray = msg:getheadervalues("diversion ) local DiversCount = #DiversArray if DiversCount > 1 then for I = 1, (DiversCount - 1) do -- Get all Diversion Headers -- Number of Diversion Headers -- Only if there s more than one -- Remove all but last header msg:removeheadervalue("diversion", DiversArray[I]) -- remove a Diversion Header end end 78
79 Completed Script M = {} function M.outbound_INVITE(msg) -- Process outbound INVITES to VM local DiversArray = msg:getheadervalues("diversion ) -- Get all Diversion Headers local DiversCount = #DiversArray if DiversCount > 1 then for I = 1, (DiversCount - 1) do -- Number of Diversion Headers in Invite -- Only if there s more than one -- Remove all but last header msg:removeheadervalue("diversion", DiversArray[I]) -- remove a Diversion Header end end end return M 79
80 Deploy the script to Unified CM Now that we have a script, what do we do with it? Apply it to the voice mail SIP trunk in Unified CM: 1. Add a SIP Normalization Script 2. Can be imported from a text file or copy/paste 3. Save the Script 4. Apply the Script to the appropriate SIP trunk 80
81 Add a SIP Normalization Script 81
82 Import the Script 82
83 Configure and save the Script 83
84 Apply the Script to the SIP trunk 84
85 Verify that your script fixes the problem INVITE SIP/2.0 Via: SIP/2.0/TCP :5060;branch=z9hG4bK12b5cc229a69621 From: "PSTN" To: Date: Wed, 19 Dec :45:01 GMT Call-ID: Supported: 100rel,timer,resource-priority,replaces User-Agent: Cisco-CUCM8.5 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Call-Info: <sip: :5060>;method="notify;event=telephone-event;duration=500" Cisco-Guid: Session-Expires: 1800 Diversion: Contact: Content-Length: 0 Call now goes to x1100 greeting (1 Diversion Header) 85
86 Some Common Scripting Forms 86
87 Navigating Script Formats Sample Normalization Scripts are available Can be confusing, not all scripts follow same format Scripts follow two basic formats: Function Name made up of the direction and SIP method example: function M.inbound_INVITE(msg) Function Name indicates purpose, but not direction or SIP Method example: function add_reply_to_header(msg) 87
88 Name = Direction + Method M = {} function M.inbound_INVITE(msg)... (function code)... end return M Good: Know exactly what SIP messages will be acted on Bad: Cannot reuse the function for anything else Need to repeat the code to duplicate processing on outbound INVITE 88
89 Example: Name with Purpose M = {} local function add_reply_to_header(msg)... (function code)... end M.outbound_INVITE = add_reply_to_header return M 89
90 Name with Purpose Good: Function name describes exactly what it does Can reuse the function without repeating it To use add_reply_to_header for inbound INVITES: M.inbound_INVITE = add_reply_to_header To use add_reply_to_header for outbound INVITES: M.outbound_INVITE = add_reply_to_header Bad: Might search through entire script to figure out the direction and SIP Method 90
91 Using Script Parameters SIP T&N allows you to set script parameters Available in Unified CM Admin pages Allows deployment of a generic script Provide script with site settings at load time Use same script in multiple locations without rewriting Example: local mydomain = scriptparameters.getvalue("localdomain") local fqdn = host..... mydomain 91
92 Setting a Script Parameter Access Unified CM Admin Set on the SIP Trunk Configuration page Must know the Parameter Name from the script Parameters are not indexed by Unified CM No pick-list provided 92
93 Enabling Tracing in Scripts Scripts can write trace information to Unified CM logs Tracing must be enabled in BOTH: Your script Unified CM Configuration Can embed tracing in every script: Use for testing and troubleshooting Disable from Unified CM Admin in production to optimize performance Trace output added to Unified CM SDI Trace 93
94 Adding Tracing to Your Scripts trace.enable() function M.inbound_NOTIFY(msg) local callid = msg:getheader("call-id") trace.format("m.inbound_notify: callid is '%s'", callid) trace.format(" -- missing URI host, no changes made") 94
95 Enabling Tracing from Unified CM 95
96 Case Study 2 96
97 MWI Lights Don t Light Adapting SIP Notify Problem: Unity Connection provides voic for user on multiple vendor PBXs Unity Connection homed to Unified CM SME SME connects to other PBXs via SIP One vendors PBX fails to update MWI status Initial Troubleshooting: Gathered traces: SIP Notify being sent to system Opened case with vendor Software release running on system needs Message-Account in the Notify 97
98 First Try Add a Message-Account header No detail available on what Notify should look like You need a Message-Account Header Make it look like the To: Header That s easy Use geturi(to) gives me the URI from the To: Use that to add a new header to message: addheader( Message-Account, <uri from geturi>) Turns out it wasn t so easy 98
99 Revisiting the Problem Discover that Message-Account isn t a SIP Header at least for this release Message-Account should be in the content body Oh, and by the way Can you make sure it is right after the Messages-Waiting line That s a bit more complicated 99
100 Tackling the Problem Can use the geturi( To ) function to grab the To: header like before You can t edit the content body in place: Read the body and save it: getcontentbody() Delete the current body: removecontentbody() Add updated body: addcontentbody() Easy: Add the Message-Account: line to beginning or end of body Harder content body processing: Add Message-Account: line right after Messages-Waiting: line Don t know where in the body Messages-Waiting: line is Maintain CR-LF terminator on each line of body Use string matching and (partial) substitution 100
101 Updating the Content Body Before and After Content-Body from Unity Connection: Messages-Waiting: yes Voice-Message: 5/0 (0/0) Fax-Message: 0/0 (0/0) Content-Body desired: Messages-Waiting: yes Message-Account: Voice-Message: 5/0 (0/0) Fax-Message: 0/0 (0/0) 101
102 Build the Script Basics M = {} function M.outbound_NOTIFY(msg)... end return M 102
103 Create the Message-Account Line -- Get the URI from To: and extract user and host local uristring = msg:geturi("to") local nuri = siputils.parseuri(uristring) nuser = nuri:getuser() nhost = nuri:gethost() -- Build the Message-Account line ma = "Message-Account: sip:".. nhost ma = ma.. "\r\n" 103
104 Update the Content Body -- Set the content type of the body local ct = "application/simple-message-summary" -- Get the existing content body local cb = msg:getcontentbody(ct) -- Build a new content body local ncb = string.gsub(cb, "Messages%-Waiting%:%s%w+%c+", "%0".. ma) -- Have to remove the existing content body and re-add msg:removecontentbody(ct) msg:addcontentbody(ct, ncb) 104
105 Taking a Look at the String Operation Breaking down the key string operation string.gsub(cb, "Messages%-Waiting%:%s%w+%c+", "%0".. ma) string.gsub takes three arguments: 1. String you want to process 2. String you want to find (match) 3. What you want to replace the matched string with Looking for Messages-Waiting: yes\r\n (or no) Take what we matched: %0 concatenate on the Message-Address we built before get the two lines we want for the Content Body Apply to the SIP trunk going to the offending PBX 105
106 Lua Demo Testing Your Lua Script Locally Processing Message-Account Test Message 106
107 A Brief Look at Other Sample Scripts 107
108 Two Additional Normalization Scripts Set-Silence Modifies SDP to set Silence Suppression off Add-Reply Adds a Header to the SIP INVITE 108
109 SDP Example: Set Silence Suppression M = {} local function M.outbound_INVITE(msg) local sdp = msg:getsdp() if sdp then sdp = sdp:gsub("a=rtpmap:8 PCMA/8000", "a=rtpmap:8 PCMA/8000\r\na=silenceSupp:off ") msg:setsdp(sdp) end end return M 109
110 Add Header Example: Add Reply-To Header M = {} local top_level_domain = scriptparameters.getvalue("top-level-domain") local function add_reply_to_header(msg) if not top_level_domain then return end local rpid = msg:getheader("remote-party-id") if not rpid then return end local replacement = string.format("<sip:%s@%s>", "%1", top_level_domain) local reply_to = rpid:gsub("<sip:(.*)@[^>]*>.*", replacement) if reply_to then msg:addheader("reply-to", reply_to) end end M.outbound_INVITE = add_reply_to_header return M 110
111 Conclusion 111
112 Some Final Thoughts If you can identify the problem, you can fix it Traces and packet captures are your friend All normalization scripts have same beginnings Just need a few Lua basics Test, write, test, fix, test, then go to production 112
113 Resources Use these for additional details SIP Chapter in Unified CM System Guide: BK_SE5FCFB6_00_cucm-system-guide-100_chapter_ html Developer Guide for SIP Transparency and Normalization Cisco Interoperability Portal Cisco Developer Network 113
114 Resources: Where to Ask Questions Cisco Developer Network has a SIP Transparency and Normalization Forum Part of the SIP Developer Portal community/-/message_boards/ category/ Post questions there and interact with other developers 114
115 Related / Recommended Sessions Milan 2014 Session Number Session Name Speaker Day BRKUCC-2932 Troubleshooting SIP with Cisco Unified Communications Paul Giralt Tuesday BRKUCC-2934 Implementation and Management of Cisco's Enterprise Session Border Controller - Cisco Unified Border Element Darryl Sladden Tuesday BRKCOL-2020 Cisco Interoperability with Microsoft Tobias Neumann Tuesday BRKUCC-2801 Cisco Expressway at the Collaboration Edge design session Kevin Roarty Tuesday BRKUCC-2008 Enterprise Dial Plan Fundamentals Johannes Krohn Wednesday BRKUCC-2340 Best practices to enable rich-media Collaboration between businesses Viraj Raut Wednesday BRKUCC-2501 Cisco UC Manager Security Kevin Roarty Wednesday Fixing SIP Problems with Cisco Unified Communications Manager's SIP Normalization Tools Mark Stover Thursday BRKUCC-3000 Advanced Dial Plan Design for Unified Communications Networks Johannes Krohn Thursday BRKUCC-2006 SIP Trunk design and deployment in Enterprise UC networks Anthony Mulchrone Friday 115
116 Questions? Thanks for Attending! I will be in the Meet the Engineer area for walk-in meetings: Thursday from 5-6 Other times by appointment through the MTE Scheduler 116
117 Key References for SIP T&N 117
118 Recommended Reading for 118
119 Call to Action Visit the World of Solutions:- Cisco Campus Walk-in Labs Technical Solutions Clinics Meet the Engineer Lunch Time Table Topics, held in the main Catering Hall Recommended Reading: For reading material and further resources for this session, please visit 119
120 Complete Your Online Session Evaluation Give us your feedback and you could win fabulous prizes. Winners announced daily. Receive 20 Cisco Daily Challenge points for each session evaluation you complete. Complete your session evaluation online now through either the mobile app or internet kiosk stations. Maximize your Cisco Live experience with your free Cisco Live 365 account. Download session PDFs, view sessions on-demand and participate in live activities throughout the year. Click the Enter Cisco Live 365 button in your Cisco Live portal to log in. 120
121
Fixing SIP Problems with UC Manager's SIP Normalization Tools
Fixing SIP Problems with UC Manager's SIP Normalization Tools Mark Stover Agenda Why have this session? Brief review of SIP When things don t work Overview of SIP Transparency and Normalization Overview
More informationFixing SIP Problems with UC Manager & CUBE Normalization Tools
Fixing SIP Problems with UC Manager & CUBE Normalization Tools Mark Stover, CCIE #6901 Consulting Systems Engineer BRKCOL-2455 Why have this session? More systems than ever use SIP Last count was 107 Products
More informationFixing SIP Problems with UC Manager's SIP Normalization Tools
Fixing SIP Problems with UC Manager's SIP Normalization Tools Mark Stover, CCIE #6901 Collaboration Consulting SE Why have this session? More systems than ever use SIP Last count was 107 Products on SIP
More informationFixing SIP Problems with UC Manager & CUBE Normalization Tools
Fixing SIP Problems with UC Manager & CUBE Normalization Tools Mark Stover, CCIE #6901 Consulting Systems Engineer Agenda Introduction (Very) Brief Review of SIP When Things Don t Work Overview of SIP
More informationSIP Reliable Provisional Response on CUBE and CUCM Configuration Example
SIP Reliable Provisional Response on CUBE and CUCM Configuration Example Document ID: 116086 Contributed by Robin Cai, Cisco TAC Engineer. May 16, 2013 Contents Introduction Prerequisites Requirements
More informationManipulating the Request or Response line. getrequestline() returns the method, request-uri, and version
CHAPTER 3 SIP s APIs The Lua scripting environment provides a set of APIs that allows messages to be manipulated These APIs are explained under the following categories: Manipulating the Request or Response
More informationDomain-Based Routing Support on the Cisco UBE
First Published: June 15, 2011 Last Updated: July 22, 2011 The Domain-based routing feature provides support for matching an outbound dial peer based on the domain name or IP address provided in the request
More informationSIP Trunk design and deployment in Enterprise UC networks
SIP Trunk design and deployment in Enterprise UC networks Tony Mulchrone Technical Marketing Engineer Cisco Collaboration Technology Group Objectives of this session a) Provide a quick overview of SIP
More informationApplication Notes for Configuring SIP Trunking between Cincinnati Bell Any Distance evantage and Avaya IP Office Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between Cincinnati Bell Any Distance evantage and Avaya IP Office Issue 1.0 Abstract These Application Notes describe
More informationSIP Transparency. Supported Features CHAPTER
CHAPTER 10 Cisco Unified Communications Manager (Unified CM) is a Back to Back User Agent (B2BUA). Therefore, any SIP to SIP call consists of 2 SIP dialogs. It is often useful to pass information from
More informationFigure 1: Incoming and Outgoing messages where SIP Profiles can be applied
Session Initiation Protocol (SIP) profiles change SIP incoming or outgoing messages so that interoperability between incompatible devices can be ensured. SIP profiles can be configured with rules to add,
More informationTroubleshooting Cisco Unity Connection
Troubleshooting Cisco Unity Connection Bryan Shapess 2 Agenda Troubleshooting Methodology Troubleshooting Tools CUC Traces Case Studies Message Waiting Indicator Visual Voicemail Single Inbox Message Delivery
More informationFigure 1: Incoming and Outgoing messages where SIP Profiles can be applied
Session Initiation Protocol (SIP) profiles change SIP incoming or outgoing messages so that interoperability between incompatible devices can be ensured. SIP profiles can be configured with rules to add,
More informationSIP Trunk design and deployment in Enterprise UC networks
SIP Trunk design and deployment in Enterprise UC networks BRKUCC-2006 Tony Mulchrone Technical Marketing Engineer Cisco Collaboration Technology Group Housekeeping We value your feedback- don't forget
More informationMid-call Re-INVITE/UPDATE Consumption
The Mid-call Re-INVITE/UPDATE consumption feature helps consume unwanted mid-call Re-INVITEs/UPDATEs locally avoiding interoperability issues that may arise due to these Re-INVITES. Feature Information
More informationSIP Trunk design and deployment in Enterprise UC networks
SIP Trunk design and deployment in Enterprise UC networks BRKUCC-2006 Tony Mulchrone Technical Marketing Engineer Cisco Collaboration Technology Group Housekeeping We value your feedback- don't forget
More informationSIP Core SIP Technology Enhancements
SIP Core SIP Technology Enhancements This feature contains the following sections: Information About SIP Core SIP Technology Enhancements, page 104 Prerequisites for SIP Core SIP Technology Enhancements,
More informationCisco Unified Communications Manager Trunks
CHAPTER 2 A trunk is a communications channel on Cisco Unified Communications Manager (Cisco Unified CM) that enables Cisco Unified CM to connect to other servers. Using one or more trunks, Cisco Unified
More informationINTERFACE SPECIFICATION SIP Trunking. 8x8 SIP Trunking. Interface Specification. Version 2.0
8x8 Interface Specification Version 2.0 Table of Contents Introduction....3 Feature Set....3 SIP Interface....3 Supported Standards....3 Supported SIP methods....4 Additional Supported SIP Headers...4
More informationTSM350G Midterm Exam MY NAME IS March 12, 2007
TSM350G Midterm Exam MY NAME IS March 12, 2007 PLEAE PUT ALL YOUR ANSWERS in a BLUE BOOK with YOUR NAME ON IT IF you are using more than one blue book, please put your name on ALL blue books 1 Attached
More informationAbstract. Avaya Solution & Interoperability Test Lab
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between Global Crossing Voice over IP services including VoIP On- Net Plus, VoIP Outbound, VoIP Local Service,
More informationAbstract. Avaya Solution & Interoperability Test Lab
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between Sotel IP Services SIP Edge Advanced SIP Trunking Solution and an Avaya IP Office Telephony Solution Issue
More informationDepartment of Computer Science. Burapha University 6 SIP (I)
Burapha University ก Department of Computer Science 6 SIP (I) Functionalities of SIP Network elements that might be used in the SIP network Structure of Request and Response SIP messages Other important
More informationSetting up Alcatel 4400 Digital PIMG Integration
up Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection Up an Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection, on page 1 Up an Alcatel 4400 Digital PIMG Integration with
More informationSetting Up an Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection
up Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection Up an Alcatel 4400 Digital PIMG Integration with Cisco Unity Connection, page 1 Up an Alcatel 4400 Digital PIMG Integration with Cisco
More informationSetting Up a Mitel SX-2000 Digital PIMG Integration with Cisco Unity Connection
Up a Mitel SX-2000 Digital PIMG Integration with Cisco Unity Connection Up a Mitel SX-2000 Digital PIMG Integration, page 1 Up a Mitel SX-2000 Digital PIMG Integration Task List for Mitel SX-2000 PIMG
More informationTechnical specifications for connecting SIP PBX to the Business Trunk service by Slovak Telekom without registration, with static routing.
Technical specifications for connecting SIP PBX to the Business Trunk service by Slovak Telekom without registration, with static routing Author: Peter Hecht Valid from: 1st January, 2019 Last modify:
More informationCCIE Collaboration.
CCIE Collaboration Cisco 400-051 Dumps Available Here at: /cisco-exam/400-051-dumps.html Enrolling now you will get access to 605 questions in a unique set of 400-051 dumps Question 1 Refer to the exhibit.
More informationVoice over IP Consortium
Voice over IP Consortium Version 1.6 Last Updated: August 20, 2010 121 Technology Drive, Suite 2 University of New Hampshire Durham, NH 03824 Research Computing Center Phone: +1-603-862-0186 Fax: +1-603-862-4181
More informationAvaya IP Office 4.1 SIP Customer Configuration Guide For use with AT&T IP Flexible Reach. Issue th April 2008
Avaya IP Office 4.1 SIP Customer Configuration Guide For use with AT&T IP Flexible Reach Issue 3.0 4 th April 2008 trademark rights, and all such rights are reserved. Page 1 of 23 Table of contents 1 Introduction...
More informationSession Initiation Protocol (SIP) Overview
Session Initiation Protocol (SIP) Overview T-110.7100 Applications and Services in Internet 6.10.2009 Jouni Mäenpää NomadicLab, Ericsson Contents SIP introduction, history and functionality Key concepts
More informationSetting Up an Avaya Definity ProLogix Digital PIMG Integration with Cisco Unity Connection
CHAPTER 4 Setting Up an Avaya Definity ProLogix Digital PIMG Integration with Cisco Unity Connection For detailed instructions for setting up an Avaya Definity ProLogix digital PIMG integration with Cisco
More informationApplication Notes for IntelePeer CoreCloud SIP Trunking Service with Avaya IP Office Release Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for IntelePeer CoreCloud SIP Trunking Service with Avaya IP Office Release 8.1 - Issue 1.0 Abstract These Application Notes describe the procedures
More informationChapter 3: IP Multimedia Subsystems and Application-Level Signaling
Chapter 3: IP Multimedia Subsystems and Application-Level Signaling Jyh-Cheng Chen and Tao Zhang IP-Based Next-Generation Wireless Networks Published by John Wiley & Sons, Inc. January 2004 Outline 3.1
More informationApplication Notes for Configuring Avaya IP Office 8.1 with Etisalat SIP Trunk service Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for Configuring Avaya IP Office 8.1 with Etisalat SIP Trunk service Issue 1.0 Abstract These Application Notes describe the procedures for configuring
More informationApplication Scenario 1: Direct Call UA UA
Application Scenario 1: Direct Call UA UA Internet Alice Bob Call signaling Media streams 2009 Jörg Ott 1 tzi.org INVITE sip:bob@foo.bar.com Direct Call bar.com Note: Three-way handshake is performed only
More informationSession Initiation Protocol (SIP) Overview
Session Initiation Protocol (SIP) Overview T-110.7100 Applications and Services in Internet 5.10.2010 Jouni Mäenpää NomadicLab, Ericsson Research Contents SIP introduction, history and functionality Key
More informationNative Call Queueing Enhancement in CUCM 11.5
Native Call Queueing Enhancement in CUCM 115 Contents Introduction Components Used Background Information Feature Overview Configuration H225 Trunk (Gatekeeper Controlled) Inter-Cluster Trunk (Non-Gatekeeper
More informationSIP Tutorial. Leonid Consulting V1.4. Copyright Leonid Consulting, LLC (2007) All rights reserved.
SIP Tutorial Leonid Consulting V1.4 Contents Contents... 2 Tables and Diagrams... 3 Introduction... 4 SIP... 5 A Brief Introduction... 5 What is SIP?... 5 Who maintains SIP?... 5 What are the elements
More informationApplication Notes for Configuring SIP Trunking between Bandwidth.com SIP Trunking Solution and an Avaya IP Office Telephony Solution Issue 1.
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between Bandwidth.com SIP Trunking Solution and an Avaya IP Office Telephony Solution Issue 1.0 Abstract These
More information2015/04/13 11:41 1/22 UNIVERGE 3C
2015/04/13 11:41 1/22 UNIVERGE 3C UNIVERGE 3C Requirements DuVoice 5.20.040 or above. Dialogic HMP. UNIVERGE 3C version 8.5.3 or above. Features Name display change. Class of service change. Do not disturb
More informationTSIN02 - Internetworking
Lecture 8: SIP and H323 Litterature: 2004 Image Coding Group, Linköpings Universitet Lecture 8: SIP and H323 Goals: After this lecture you should Understand the basics of SIP and it's architecture Understand
More information2018/05/18 23:05 1/2 UNIVERGE 3C
2018/05/18 23:05 1/2 UNIVERGE 3C Table of Contents UNIVERGE 3C... 1 Requirements... 1 Features... 1 PBX Configuration Part One... 1 Active Directory... 1 Web Services User... 1 SIP User Agents... 4 Class
More informationConfigure Jabber Extend and Connect and Modify Calling Party Display
Configure Jabber Extend and Connect and Modify Calling Party Display Contents Introduction Prerequisites Requirements Components Used Configure Network Diagram Troubleshooting example Introduction This
More informationConfiguring SIP MWI Features
This module describes message-waiting indication (MWI) in a SIP-enabled network. Finding Feature Information, on page 1 Prerequisites for SIP MWI, on page 1 Restrictions for SIP MWI, on page 2 Information
More informationBRKCOC-2399 Inside Cisco IT: Integrating Spark with existing large deployments
Inside Cisco IT: Integrating Spark with existing large deployments Jan Seynaeve, Sr. Collaborations Engineer Luke Clifford, Sr. Collaborations Engineer Cisco Spark How Questions? Use Cisco Spark to communicate
More informationAbstract. Avaya Solution & Interoperability Test Lab
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between the PAETEC Broadsoft based SIP Trunking Solution and an Avaya IP Office Telephony Solution Issue 1.0 Abstract
More informationConfigure Selective Based Workflow for Incoming Calls on Finesse
Configure Selective Based Workflow for Incoming Calls on Finesse Contents Introduction Prerequisites Requirements Components Used Configure Configurations CUCM Configuration MediaSense Configuration UCCX
More informationEnterprise Voice SUBSCRIBER GUIDE
Enterprise Voice SUBSCRIBER GUIDE Conterra Networks Enterprise Voice SUBSCRIBER GUIDE 3 TABLE OF CONTENTS Table of Contents Introduction... 6 Logging in... 6 Navigation Bar, Sub-Menu and Page Layout...
More informationa. Draw a network diagram, showing how a telephone in the US would make calls to a telephone on Deception Island. (15 points).
TSM 350 IP Telephony Fall 2004 E Eichen Exam 1 (Midterm): November 10 Solutions 1 True or False: a Call signaling in a SIP network is routed on a hop-by-hop basis, while call signaling in an H323 network
More informationDeploy Webex Video Mesh
Video Mesh Deployment Task Flow, on page 1 Install Webex Video Mesh Node Software, on page 2 Log in to the Webex Video Mesh Node Console, on page 4 Set the Network Configuration of the Webex Video Mesh
More informationApplication Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution Issue 1.
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between McLeodUSA SIP Trunking Solution and an Avaya IP Office Telephony Solution Issue 1.1 Abstract These Application
More informationMultimedia networking: outline
Multimedia networking: outline 9.1 multimedia networking applications 9.2 streaming stored video 9.3 voice-over-ip 9.4 protocols for real-time conversational applications: SIP Skip RTP, RTCP 9.5 network
More informationSetting Up a Serial (SMDI, MCI, or MD-110) PIMG Integration with Cisco Unity Connection
CHAPTER 11 Setting Up a Serial (SMDI, MCI, or MD-110) PIMG Integration with Cisco Unity Connection For detailed instructions for setting up a serial (SMDI, MCI, or MD-110) PIMG integration with Cisco Unity
More informationICE / TURN / STUN Tutorial
BRKCOL-2986 ICE / TURN / STUN Tutorial Kristof Van Coillie, Technical Leader, Services Cisco Spark How Questions? Use Cisco Spark to communicate with the speaker after the session 1. Find this session
More informationCisco TelePresence Integration Guide Documentation for integrating Cisco CTS/TX TelePresence Systems with BlueJeans
Cisco TelePresence Integration Guide Documentation for integrating Cisco CTS/TX TelePresence Systems with BlueJeans Last Updated: April 2018 5 1 6 C l y d e A v e n u e M o u n t a i n V i e w, C A 9 4
More informationSetup for Cisco Unified Communications Manager
Setup for Cisco Unified Communications Manager This chapter describes how you can set up Cisco Jabber for ipad using Cisco Unified Communications Manager. System and Network Requirements, page 1 Recommended
More informationSignaling trace on GSM/CDMA VoIP Gateway
Signaling trace on GSM/CDMA VoIP Gateway Part1. Login the gateway & General Knowledge the command This is a document for some customers who need to get the logs on gateway Tips: The document is fit for
More informationCisco Unified CM SIP Trunking, Session Management, and Global Dial Plan Replication
LTRUCC-2150 Cisco Unified CM SIP Trunking, Session Management, and Global Dial Plan Replication Paul Giralt - @PaulGiralt Markus Schneider - @Markus73 Agenda Objectives Technology Overview Unified CM Session
More informationCCS-UC-1 SIP Endpoint with ShoreTel Connect System Configuration Guide Crestron Electronics, Inc.
CCS-UC-1 SIP Endpoint with ShoreTel Connect System 21.80.7840.0 Configuration Guide Crestron Electronics, Inc. Crestron product development software is licensed to Crestron dealers and Crestron Service
More informationApplication Notes for Configuring SIP Trunking between Global Crossing SIP Trunking Service and an Avaya IP Office Telephony Solution Issue 1.
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between Global Crossing SIP Trunking Service and an Avaya IP Office Telephony Solution Issue 1.0 Abstract These
More informationSIP System Features. SIP Timer Values. Rules for Configuring the SIP Timers CHAPTER
CHAPTER 4 Revised: October 30, 2012, This chapter describes features that apply to all SIP system operations. It includes the following topics: SIP Timer Values, page 4-1 Limitations on Number of URLs,
More informationOverview of SIP. Information About SIP. SIP Capabilities. This chapter provides an overview of the Session Initiation Protocol (SIP).
This chapter provides an overview of the Session Initiation Protocol (SIP). Information About SIP, page 1 How SIP Works, page 4 How SIP Works with a Proxy Server, page 5 How SIP Works with a Redirect Server,
More informationCall Park and Directed Call Park
Call Park Overview Call Park Overview, on page 1 Call Park Prerequisites, on page 2 Call Park Configuration Task Flow, on page 2 Call Park Interactions and Restrictions, on page 17 Troubleshooting Call
More informationMITEL SIP CoE Technical. Configuration Note. Configure Mitel MiVoice Office 6.1 SP1 PR2 for use with IntelePeer SIP Trunking. SIP CoE XXX
MITEL SIP CoE Technical Configuration Note Configure Mitel MiVoice Office 6.1 SP1 PR2 for use with IntelePeer SIP Trunking SIP CoE 12-4940-00XXX NOTICE The information contained in this document is believed
More informationUnderstanding SIP exchanges by experimentation
Understanding SIP exchanges by experimentation Emin Gabrielyan 2007-04-10 Switzernet Sàrl We analyze a few simple scenarios of SIP message exchanges for a call setup between two SIP phones. We use an SIP
More informationMohammad Hossein Manshaei 1393
Mohammad Hossein Manshaei manshaei@gmail.com 1393 Voice and Video over IP Slides derived from those available on the Web site of the book Computer Networking, by Kurose and Ross, PEARSON 2 Multimedia networking:
More informationN-Squared Software SIP Specialized Resource Platform SIP-SDP-RTP Protocol Conformance Statement. Version 2.3
N-Squared Software SIP Specialized Resource Platform SIP-SDP-RTP Protocol Conformance Statement Version 2.3 1 Document Information 1.1 Scope and Purpose This document describes the implementation of the
More informationExtensions to Session Initiation Protocol (SIP) and Peer-to-Peer SIP
Extensions to Session Initiation Protocol (SIP) and Peer-to-Peer SIP T-110.7100 Applications and Services in Internet 1.10.2008 Jouni Mäenpää NomadicLab, Ericsson Contents Extending SIP SIP extension negotiation
More informationControlONE Technical Guide
ControlONE Technical Guide Recording Interface - SIPREC v6.1 1 of 9 Introduction 3 Definitions 3 Interface Description 3 Session Flow 3 Call Information 4 Media Session 5 Security 5 Licensing 5 Examples
More informationSIP System Features. Differentiated Services Codepoint CHAPTER
CHAPTER 6 Revised: December 30 2007, This chapter describes features that apply to all SIP system operations. It includes the following topics: Differentiated Services Codepoint section on page 6-1 Limitations
More informationSIP Standard Line Interface
This chapter describes the external interface for Cisco Unified CM SIP line-side devices. It highlights SIP primitives that are supported on the line-side interface and describes call flow scenarios that
More informationApplication Note 3Com VCX Connect with SIP Trunking - Configuration Guide
Application Note 3Com VCX Connect with SIP Trunking - Configuration Guide 28 May 2009 3Com VCX Connect Solution SIP Trunking Table of Contents 1 3COM VCX CONNECT AND INGATE... 1 1.1 SIP TRUNKING SUPPORT...
More informationSIP Pass Through APIs
CHAPTER 5 Cisco Unified CM is a Business to Business User Application (B2BUA) with respect to SIP call processing The pass through object provides a set of APIs that allows information to be passed from
More informationApplication Notes for Configuring the ADTRAN NetVanta UC Server with Avaya IP Office 6.1 Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for Configuring the ADTRAN NetVanta UC Server with Avaya IP Office 6.1 Issue 1.0 Abstract These Application Notes describe the procedure for
More informationApplication Notes for Configuring SIP Trunking between TelePacific SmartVoice SIP Connect and an Avaya IP Office Telephony Solution 1.
Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunking between TelePacific SmartVoice SIP Connect and an Avaya IP Office Telephony Solution 1.0 Abstract These Application
More informationITBraindumps. Latest IT Braindumps study guide
ITBraindumps http://www.itbraindumps.com Latest IT Braindumps study guide Exam : 300-075 Title : Implementing Cisco IP Telephony & Video, Part 2 v1.0 Vendor : Cisco Version : DEMO Get Latest & Valid 300-075
More informationENTERPRISE SUBSCRIBER GUIDE
ENTERPRISE SUBSCRIBER GUIDE Enterprise Subscriber Guide 880 Montclair Road Suite 400 Birmingham, AL 353 www. TABLE OF CONTENTS Table of Contents Introduction...6 Logging In...6 Navigation Bar, Sub-Menu
More informationCisco Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) For More Information - Visit:
CertsChief Guaranteed Success with Accurate & Updated Questions. Cisco 300-075 Implementing Cisco IP Telephony and Video, Part 2 (CIPTV2) Questions & Answers PDF For More Information - Visit: https://www.certschief.com/
More informationGSM VoIP Gateway Series
VoIP Gateway Series SIP Protocol Debugging Service Overview www.addpac.com AddPac Technology Sales and Marketing Contents? Network Diagram for SIP Debugging? SIP Debugging Access Method via Console Port?
More informationApplication Notes for Windstream SIP Trunking Service using Broadsoft Platform with Avaya IP Office Issue 1.0
Avaya Solution & Interoperability Test Lab Application Notes for Windstream SIP Trunking Service using Broadsoft Platform with Avaya IP Office 8.1 - Issue 1.0 Abstract These Application Notes describe
More informationContents XO COMMUNICATIONS CONFIDENTIAL 1
www.xo.com XO SIP Service Customer Configuration Guide for Cisco Unified Communications Manager (CUCM) 8.0.3 XO SIP Packages 1 and 2, implemented without Cisco Unified Border Control Element (CUBE) SIP
More informationCall Forwarding. Call Forwarding Overview
Overview, page 1 Configuration Task Flow, page 3 Interactions and Restrictions, page 12 Overview As a user, you can configure a Cisco Unified IP Phone to forward calls to another phone. The following call
More informationIn Depth Analysis of Ringback for all VoIP and Analog Protocols
In Depth Analysis of Ringback for all VoIP and Analog Protocols Contents Introduction Prerequisites Requirements Components Used Background Information Protocols ISDN Q.931 (T1 / E1 / BRI) H.323 SIP MGCP
More informationVoice over IP (VoIP)
Voice over IP (VoIP) David Wang, Ph.D. UT Arlington 1 Purposes of this Lecture To present an overview of Voice over IP To use VoIP as an example To review what we have learned so far To use what we have
More informationConfiguring Triggers. Viewing and Deleting Triggers
Configuring Triggers Viewing and Deleting Triggers Adding a Trigger Viewing, Adding, Moving, and Deleting Rules for a Trigger Adding, Editing, and Deleting Conditions for a Trigger Rule Viewing and Deleting
More informationConfiguration Guide For Use with AT&T s IP Flexible Reach Service. Version 1/Issue 7. July 30, 2008
Configuration Guide For Use with AT&T s IP Flexible Reach Service Version 1/Issue 7 July 30, 2008 Page 1 of 42 TABLE OF CONTENTS 1 Introduction... 4 2 Version Information... 6 3 Special Notes... 7 ShoreTel
More informationSIP (Session Initiation Protocol)
Stanford University Electrical Engineering EE384B - Mutimedia Networking and Communications Group #25 SIP (Session Initiation Protocol) Venkatesh Venkataramanan Matthew Densing
More informationCisco Unified Border Element Intercluster Lookup Service
Cisco Unified Border Element Intercluster Lookup Service The Cisco Unified Border Element (Cisco UBE) Intercluster Lookup Service feature enables Cisco Unified Communications Manager to establish calls
More informationSIP profile setup. About SIP profile setup. SIP profile reset. SIP profile deletion
SIP profile setup This chapter provides information to configure and locate SIP profiles. A SIP profile comprises the set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles
More informationAbstract. Avaya Solution & Interoperability Test Lab
Avaya Solution & Interoperability Test Lab Configuring SIP trunks between Avaya Aura Session Manager Release 6.2, Avaya Meeting Exchange Enterprise Edition Release 6.2 and Cisco Unified Communications
More informationNext Generation Mobile Collaboration
Next Generation Mobile Collaboration PSOUCC-2777 Chris Wiborg Director, Cisco Collaboration Portfolio Marketing @cwiborg Agenda Why Mobile Collaboration Matters Shifting User Expectations Delivering Value
More informationTechnical specifications for connecting SIP PBX to the Business Trunk service by Slovak Telekom.
Technical specifications for connecting SIP PBX to the Business Trunk service by Slovak Telekom Author: Peter Hecht Valid from: September, 2015 Version: 70 1 Use of the service Service Business Trunk is
More informationConfiguration Example for CUCM Non-Secure SIP Integration with CUC
Configuration Example for CUCM Non-Secure SIP Integration with CUC Contents Introduction Prerequisites Requirements Components Used Configure Configuration on CUCM Configuration on Unity Connection Verify
More informationTelephony Integration
Introduction, page 1 Phone System, page 2 Port, page 5 Port Group, page 6 Trunk, page 12 Speech Connect Port, page 13 Audio and Video Format Using Phone, page 14 Security, page 15 IPv6 in Unity Connection
More informationSIPPING Working Group A. Johnston, Ed. Internet-Draft Avaya Intended status: BCP R. Sparks Expires: January 12, 2009 Estacado Systems C. Cunningham S.
SIPPING Working Group A. Johnston, Ed. Internet-Draft Avaya Intended status: BCP R. Sparks Expires: January 12, 2009 Estacado Systems C. Cunningham S. Donovan Cisco Systems K. Summers Sonus July 11, Status
More informationSIP Protocol Debugging Service
VoIP Gateway Series SIP Protocol Debugging Service Overview www.addpac.com AddPac Technology 2011, Sales and Marketing Contents Network Diagram for SIP Debugging SIP Debugging Access Method via Console
More informationMultiparty Conferencing for Audio, Video and Web Collaboration using Cisco Meeting Server
Multiparty Conferencing for Audio, Video and Web Collaboration using Cisco Meeting Server Paul Giralt (pgiralt@cisco.com) Markus Schneider (marschne@cisco.com) LTRCOL-2250 Agenda Cisco Meeting Server Overview
More informationConfiguring Multi-Tenants on SIP Trunks
The feature allows specific global configurations for multiple tenants on SIP trunks that allow differentiated services for tenants. allows each tenant to have their own individual configurations. The
More informationAbstract. Avaya Solution & Interoperability Test Lab
Avaya Solution & Interoperability Test Lab Application Notes for Uecomm/Optus Evolve SIP Trunking Service with Avaya IP Office 9.1.6 and Avaya Session Border Controller for Enterprise 7.0 - Issue 1.0 Abstract
More information