Internet Engineering Task Force SIP WG Internet Draft Ilya Slain draft-ietf-slain-sip-config-parameters-00 Bich Nguyen March 22, 2001 Cisco Systems, Inc. Expires: August 2001 Configuration Parameters for IP Telephony End Systems STATUS OF THIS MEMO This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress". The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt To view the list Internet-Draft Shadow Directories, see http://www.ietf.org/shadow.html. Abstract Recently the alternatives for configuring SIP IP telephony devices have been proposed. This document proposes the list of common configuration parameters to be supplied to those devices as part of their configuration. Slain [Page 1] Internet Draft SIP March 22, 2001 1 Introduction Previous work [1] has identified general approaches to a standardized vendor-independent SIP IP endpoint configuration mechanism. Here we attempt to fulfill a much more narrow task to identify the set of SIP configuration parameters that will be common to all SIP endpoints. The intent of this draft is not to define the specific syntax and of the configuration file, but, rather, to enumerate the parameters and their value sets. The discussion of the syntax is orthogonal and can proceed separately. For clarity, the description of parameter values is given in the BNF notation; however, it is not intended to enforce any syntactical structure. Sec. 2 contains the exhaustive list of configuration parameters, while Sec. 3 presents a logical grouping of those parameters. Throughout this draft we use the rules for the basic parsing constructs from [2], Appendix C, Sec. C.1 (e.g. "DIGIT", "unreserved", etc.) and also higher level parsing constructs from [2] (e.g. "SIP-URL", "username", etc.) 2 SIP Configuration Parameters This sections specifies the list of SIP configuration parameters. The list is broken down into functional categories. SIP configuration can be grouped into the following main sections: o Identification settings o Default destination URLs and routes o Registration parameters o Timer parameters (timeout and counters settings) o Feature settings o Media settings Slain [Page 2] Internet Draft SIP March 22, 2001 2.1 Identification Settings Identification settings of the endpoint are on a per line basis, as each line represents a separately addressable SIP endpoint. The concept of a "line" is used here to identify the logical SIP endpoints at this device. Each of the endpoints can have the following parameters: o SIP-Identification-LineName The SIP URL by which it is addressable in the telephony network. Parameters: line-name Types: line-name = *(unreserved) o SIP-Identification-LineAuthenticationName Per line authentication name for SIP HTTP Digest authentication. Parameters: line-auth-name Types: line-auth-name = *(unreserved) o SIP-Identification-LineAuthenticationPassword Per line authentication password for SIP HTTP Digest authentication. Parameters: line-password Types: line-password = *(unreserved) 2.2 Default Destinations Settings Each endpoint needs to have default destination settings to complete the call without having the user specify the full addressing of the callee. o SIP-Destination-ProxyAddress Proxy address. Parameters: proxy-address Types: proxy-address = IPv4address | hostname o SIP-Destination-ProxyPort Proxy port. Parameters: proxy-port Types: proxy-port = num Slain [Page 3] Internet Draft SIP March 22, 2001 2.3 Registration Settings Settings pertaining to registration. o SIP-Registration-Enabled Indicates whether the endpoint needs to register with the proxy. Parameters: enable_flag Types: enable_flag = TRUE | FALSE o SIP-Registration-Timeout Proxy registration expiration timeout period (in seconds). Parameters: num 2.4 Timer Settings Settings pertaining to SIP timers. o SIP-Timers-TimerT1 SIP T1 timer (in milliseconds). Parameters: num o SIP-Timers-TimerT2 SIP T2 timer (in milliseconds). Parameters: num o SIP-Timers-RetransmissionMaxNumber Max number of SIP message retransmissions per transaction (all messages except the INVITE message). Parameters: num o SIP-Timers-RetransmissionMaxNumberInvite Max number of INVITE SIP message retransmissions per transaction. Parameters: num o SIP-Timers-InviteExpiration Outstanding INVITE expiration timeout period (in seconds). Parameters: num Slain [Page 4] Internet Draft SIP March 22, 2001 2.5 Features Settings Device feature settings. o SIP-Feature-* Feature configuration. Parameters: enable_flag The mechanism needs to be created for vendor registration of new feature names and feature-specific parameters. The discussion of the solution to both issues is outside the scope of this document. A sample list of features is provided here: + Call Hold + Blind Call Transfer + Attended Call Transfer + Call Conference + Call Forward Local On Busy + Call Forward Local On No Answer + Call Forwars Local Always + Anonymous Call Blocking + Caller Id Blocking + Do Not Disturb o SIP-VoiceMail Specifies settings for Voice Mail Parameters: enable_flag, voice_mail_address Types: voice_mail_address = userinfo | SIP-URL 2.6 Media Settings o SIP-Media-CodecsAdvertized The list of codecs to advertise in the invitation. Parameters: codec_list Types: codec_list = (codec_list codec) | empty codec = o SIP-Media-CodecsPreferred The ordered list of codecs in the order of preference. Parameters: codec_list o SIP-Media-DTMFTransportMethod Specifies whether the out-of-band DTMF signaling should be used during the call. Parameters: enable_flag, out_of_band_dtmf_method Slain [Page 5] Internet Draft SIP March 22, 2001 Types: out_of_band_dtmf_method = "avt" | "avt_always" The AVT method of transport refers to the DTMF transport method defined in [3]. When "avt" is specified, the AVT method will be used when both sides have indicated the support for AVT transport in the SDP. As a slight variation, when "avt_always" is specified, the endpoint will use AVT as a method for DTMF transport no matter whether the far end has indicated the support for AVT or not. 3 Logical grouping of SIP configuration parameters This section describes the logical grouping of SIP configuration parameters. Such grouping can be used for a SIP configuration file layout. All the parameters can be classified as global or line specific parameters. SIP-Config = SIP-Config-Global SIP-Config-Lines SIP-Config-Global = SIP-Timers-TimerT1 SIP-Timers-TimerT2 SIP-Timers-RetransmissionMaxNumber SIP-Timers-RetransmissionMaxNumberInvite SIP-Timers-InviteExpiration SIP-Feature-* SIP-VoiceMail SIP-Media-CodecsAdvertized SIP-Media-CodecsPreferred SIP-Media-DTMFTransportMethod SIP-Config-Lines = (SIP-Config-Lines SIP-Config-Line) | empty SIP-Config-Line = SIP-Identification-LineName SIP-Identification-LineAuthenticationName SIP-Identification-LineAuthenticationPassword SIP-Destination-ProxyAddress SIP-Destination-ProxyPort SIP-Registration-Enabled SIP-Registration-Timeout Slain [Page 6] Internet Draft SIP March 22, 2001 4 Authors' Addresses Ilya Slain Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA electronic mail: islain@cisco.com Bich Nguyen Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA electronic mail: binguyen@cisco.com 5 Bibliography [1] H. Schulzrinne, "Configuring IP Telephony End Systems", Internet Draft draft-schulzrinne-sip-config-00.txt, IETF, SIP WG, Dec. 2000. [2] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP: Session Initiation Protocol" Internet Draft draft-ietf-sip-rfc2543bis-02.ps, IETF, SIP WG, Sep. 2000. [3] H. Schulzrinne, S. Petrack, "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals", RFC 2833, IETF, May 2000. [4] H. Schulzrinne, "RTP Profile for Audio and Video Conferences with Minimal Control", RFC 1890, IETF, Jan. 1996. Slain [Page 7] Internet Draft SIP March 22, 2001 Full Copyright Statement Copyright (c) The Internet Society (2000). All Rights Reserved. This document and translations of it may be copied and furnished to others, and derivative works that comment on or otherwise explain it or assist in its implementation may be prepared, copied, published and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies and derivative works. However, this document itself may not be modified in any way, such as by removing the copyright notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or as required to translate it into languages other than English. The limited permissions granted above are perpetual and will not be revoked by the Internet Society or its successors or assigns. This document and the information contained herein is provided on an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. Slain [Page 8]