© 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v5.03-1 Deployment of Cisco Unified CallManager Release Release 5.0 Endpoints Configuring SIP Endpoints. - презентация

1 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Deployment of Cisco Unified CallManager Release Release 5.0 Endpoints Configuring SIP Endpoints

2 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Cisco Unified CallManager Release 5.0 SIP Phones Cisco Unified CallManager Release 5.0 offers SIP support for these Cisco IP phone models: Cisco Unified IP Phone 7970/71 Cisco Unified IP Phone 7960/61 Cisco Unified IP Phone 7940/41 Cisco Unified IP Phone 7911 Cisco Unified IP Phone 7906 Cisco Unified IP Phone 7905/12

3 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Cisco Unified CallManager Release 5.0 SIP Features Functions and features supported for SIP calls: Basic calls between SIP endpoints and Cisco Unified CallManager DTMF relay between SIP endpoints and Cisco Unified CallManager Supplementary services that are initiated if an MTP is allocated Ringback tone during blind transfer Supplementary services that are initiated by SIP endpoint Enhanced call identification services

4 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Gateways SIP SoftClients SIP Phones SIP Apps SIP Network Unified Messaging CTI Apps Soft Phones Conf/Xcode SCCP MGCP H.323 CTI SIP SIP Support In Cisco Unified CallManager Release 4. X Gateways DSP Resources Rich-Media Conferencing Cisco Unified CallManager Release 4. x SIP Trunk Video Endpoints SIP Proxy Server Cisco and Third-Party Phones SIP support was limited to trunk-side interface and required a proxy server to connect to the SIP network for call control.

5 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SCCP MGCP H.323 CTI SIP/SIMPLE/KPML Cisco IP Communicator Unity-Unity Connection CTI Apps Gateways MeetingPlace/ MP Express Cisco Unified Presence Server Carriers / Other PBXs Cisco and Third-Party Phones Soft Phones Video Endpoints CCME Microsoft LCS IBM Sametime SIP Support In Cisco Unified CallManager Release 5.0 Cisco Unified CallManager Release 5.0 CallManager Release 5.0 Native SIP, SIMPLE, support on both line-side and trunk-side is added for audio and video calls.

6 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Call Flow: Successful Call Setup and Disconnect Cisco SIP Phone INVITE 100 Trying 180 Ringing 200 OK ACK BYE 200 OK way RTP channel 6 AB

7 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Call Flow: Called User Is Busy Cisco SIP Phone INVITE 100 Trying 486 Busy Here ACK AB

8 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Call Flow: Simple Call Hold Cisco SIP Phone INVITE (c=IN IP ) 200 OK way RTP channel 2 AB Call setup procedure 1 ACK 5 INVITE (c=IN IP4 user B IP) 200 OK 7 8 ACK 9 RTP channel torn down 6 2-way RTP channel reestablished 10

9 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Configuring SIP Phones Provisioning a Cisco SIP phone is just like provisioning an SCCP phone. The protocol choice (SIP or SCCP) that you choose when you add the phone automatically dictates what firmware filename gets specified in the phone configuration file. Cisco Unified CallManager Device Defaults

10 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Choosing the Protocol for Auto-Registered Phones Protocol choice dictates what firmware filename gets specified in the default configuration file for each phone model. When set to SIP, only applies to phones that can run SIP. SCCP-only phone models will still auto-register using SCCP.

11 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Choosing the Protocol for a Specific Phone

12 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Profile Provides SIP-specific configuration information for SIP phones and SIP trunks Assigned to SIP phones from the Phone Configuration window Comes with a standard SIP profile that you can use, or copy and modify to create a new profile

13 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Profile Configuration Page SIP profiles provide SIP-specific phone settings such as default telephony event payload type, registration and keepalive timers, and media ports. Standard SIP Profile

14 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Phone Configuration Page Standard SIP phone button template (2 lines and 4 speed-dial buttons) Standard Common Phone Profile is the same for SIP or SCCP phones. Enabled by default.

15 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Phone Configuration Page (Cont.) SIP phone Security Profile Settings You must also choose a SIP Phone Security Profile and SIP Profile to assign to the phone. Optionally, assign SIP dial rules, if configured, and other parameters as desired.

16 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Converting Cisco SCCP Phones 7940 and 7960 to SIP 1. Download the files from Cisco SIP Phone 7940/7960 software depository. 2. Rename the file SIPDefaultGeneric.cnf to SIPDefault.cnf. 3. Rename the file SIPConfigGeneric.cnf to SIPmac_address.cnf, for each phone. 4. Reset the phones and ensure that they get IP address, gateway address, and TFTP server address.

17 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Migrating Cisco IP Phone 7970 and 7971 from SCCP to SIP BAT used to convert a Cisco IP phone that is in use from SCCP to SIP

18 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Third-Party Phone Configuration Cisco Unified CallManager Release 5.0 supports RFC compliant SIP phones from third-party companies. Two categories of third-party SIP phones in Cisco Unified CallManager: –Basic supports one line and consumes three license units. –Advanced supports up to eight lines and video, and consumes six license units. Third-party SIP phones do not send their MAC address to Cisco Unified CallManager in their REGISTER message. Therefore, you must use Digest Authentication with third-party SIP phones. Configuration is performed on Cisco Unified CallManager and on the phone itself.

19 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Digest Authentication for Phone Identification DN = 1001 AuthID = block MAC = ABC REGISTER 1001 username=block Cisco Unified CallManager Third-Party SIP Phone Device config (mac) End User config block Line config (1001) Find associated device Respond with device/line config Database Query End User config block CCM

20 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Third-Party SIP Phone Configuration Steps 1. Configure the end user in Cisco Unified CallManager. 2. Configure the device in Cisco Unified CallManager. 3. Associate the device to the end user. 4. Configure the phone with the end user ID.

21 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Third-Party Phone Configuration 1. Configure the end user. 2. Select the end user ID in Digest User drop-down list in Phone Configuration. 3. Be sure that there is only one device associated with the end user ID

22 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Third-Party Phone Configuration (Cont.) 4. Configure the authorization ID on the phone and ensure that the phone is enabled to use it. block The proxy address should be the Cisco Unified CallManager IP address or name. If the phone uses a directory number instead of an authorization ID, the User ID in the End User Configuration in Cisco Unified CallManager must also be the directory number.

23 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Impact of Dial Rules or KPML on User Experience User experience is like that of a cell phone: 1. User enters digits into keypad 2. User presses Dial softkey 3. Phone sends digits and call is placed User experience is like that of an SCCP phone: 1. User enters digits into keypad 2. When digits match a valid pattern, the phone sends digits automatically and call is placed without user having to press the Dial softkey Cisco SIP Phone with Dial Rules or KPML Third-Party or Cisco SIP Phone Without SIP Dial Rules or KPML KPML enables digit-by-digit dialing. Dial rules tell the Cisco SIP IP Phone when enough digits are collected before call processing takes place. Both enable a user to dial a destination without pressing the Dial key.

24 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Call Flow: Digit Collection Cisco Unified CallManager performs digit-by-digit collection and routes the call as soon as enough digits are collected. SCCP phone SIP Phone Using KPML SIP Phone with Local Dial Map Digit 1 Digit 0 INVITE Trying 200 OK NOTIFY (0) 200 OK INVITE 10 UNSUBSCRIBE 200 OK Cisco Unified CallManager performs digit-by-digit collection and routes the call as soon as enough digits are collected; Performance impacts on Cisco Unified CallManager in terms of call signaling bandwidth. Local dial rules are configured in Cisco Unified CallManager and downloaded to the phone; Phone collects all digits and sends them to Cisco Unified CallManager. SUBSCRIBE 100 Trying KPML can also be used for digit collection when dialing on a Cisco SIP phone. 200 OK

25 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Dial Rules and KPML Support in Cisco Unified CallManager Release 5.0 Dial rules support in Cisco Unified CallManager Release 5.0: –Local dial rules are called SIP Dial Rules and are configured in Cisco Unified CallManager Release 5.0. –Cisco 8.0 SIP phones with firmware version 8.0 download dial rules from TFTP server. –Due to syntax differences, one set of dial rules must be defined for Cisco IP Phone 7940, 7960, 7911, 7941, 7961, and 797x models, while another set for Cisco IP Phone 7905 and 7912 models. KPML support in Cisco Unified CallManager Release 5.0: –KMPL is implemented as per draft-ietf-sipping-kpml.draft-ietf-sipping-kpml –KMPL is only supported on Cisco IP Phone 7911, 7941, 7961, and 797x models; Cisco IP Phone 7905, 7912, 7940, and 7960 models must use local dial rules. –If the third-party SIP phone does not support KPML, you must use locally-defined digit maps.

26 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Dial Rules and KPML Interworking Is KPML Enabled? Are SIP Dial Rules Defined? User Must Press Dial Softkey No Yes Send En-bloc INVITE Were Enough Digits Received? Call Is Routed Yes SUBSCRIBE To KPML No Are SIP Dial Rules Defined? Were Enough Digits Received? Yes Reorder Tone No Yes Send En-bloc INVITE

27 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Dial Rules Configuration Overview Dial rules enable the Cisco SIP phone to know when enough digits are collected before call processing takes place. Dial rule syntax is slightly different on Cisco IP Phone 7905/12 models than it is on Cisco IP Phone 797x, 7941, 7961, 7911 models. Dial rule syntax varies considerably on third-Party SIP phones, and some do not support dial rules.

28 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v SIP Phone Dial Rules Configuration Sequence 1. The administrator configures the SIP dial rules and associates them with the SIP phone. 2. Cisco Unified CallManager notifies the TFTP server of the change. 3. The TFTP server rebuilds the dial rules configuration file of the phone. 4. The administrator resets or restarts the phone and the dial rules file is downloaded to the phone. TFTP Server Cisco IP Phone _DialRules DR7970_DialRules.xml Cisco Unified CallManager

29 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Dial Rule Patterns and Phone Models Supports KMPL Uses Which Dial Rule Pattern Cisco IP Phone 7905/12 No7905_7912 Cisco IP Phone 7940/60 No7940_7960_OTHER Cisco IP Phone 797x, 7941, 7961, 7911 Yes7940_7960_OTHER Even if the phone supports KMPL, the recommendation is to use SIP dial rules to increase the performance of Cisco Unified CallManager.

30 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Dial Rule Pattern Syntax Dial RulePattern FormatMeaning 7940_7960_OTHER 7905_7912 Period (.)Matches any digit Asterisk (*)Matches one or more characters Pound sign (#) Acts as the terminating key On 7905_7912 pattern, you must precede the # (or other terminating key) with a > Comma (,) Causes the phone to generate a secondary dial tone 7905_7912 Variances Hyphen (-) More digits can be entered; hyphen must appear at the end tntn Sets the timeout to n seconds Example: 6,..... will match 6, play secondary dial tone, then match any 5-digit directory number.

31 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Configuring SIP Dial Rules 1. Choose Call Routing > Dial Rules > SIP Dial Rules 2. Choose the dial pattern depending on the phone type

32 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Configuring SIP Dial Rules (Cont.) Dial parameters: Pattern = Digits Timeout = Number of seconds to wait after match to dial –0 means dial immediately User = Tag added to the dialed number. –IP and phone are valid values (not case- sensitive). –Use only for 7940_7960_OTHER dial rules.

33 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Add the SIP Dial Rules to the Phone Add SIP dial rule in the Protocol Specific Information pane of the Phone Configuration window.

34 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Summary Cisco Unified CallManager supports SIP on Cisco IP Phone 7970/71, 7960/61, 7940/41, 7911, and 7905/12 models as well as selected third-party phones. Cisco Unified CallManager integrates native SIP (RFC 2833) and SIMPLE support on both line-side and trunk-side interfaces for a variety of endpoints, gateways, and applications. External SIP proxy server is no longer required. DTMF and KPML can be used for digit collection when dialing on a SIP phone. Cisco Unified CallManager collects digits either one at a time, or, more efficiently, using SIP dial rules. Provisioning a Cisco SIP phone is just like provisioning a Cisco IP phone controlled by SCCP.

35 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v Summary (Cont.) SIP profile provides SIP-specific configuration information for SIP phones and SIP trunks. Conversion of a Cisco SCCP phone to a Cisco SIP phone is done either by manual replacement of TFTP configuration files, or by using Cisco Unified CallManager BAT. Third-party SIP phones do not send their MAC address at boot time; all the configuration is performed on Cisco Unified CallManager and on the phone itself. When you configure a SIP dial plan and associate it with a Cisco SIP phone, TFTP server builds a new set of files for the Cisco SIP phone. Once the you initiate the phone reset, the new dial rules are applied to the phone.

36 © 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v