User`s manual
Table Of Contents
- Mediant 2000 & TP-1610 & TP-260/UNI SIP User’s Manual Version 5.0
- Table of Contents
- List of Figures
- List of Tables
- Notices
- 1. Overview
- 2. Physical Description
- 3. Installation
- 4. Getting Started
- 5. Web Management
- Computer Requirements
- Protection and Security Mechanisms
- Accessing the Embedded Web Server
- Getting Acquainted with the Web Interface
- Protocol Management
- Advanced Configuration
- Status & Diagnostic
- Software Update Menu
- Maintenance
- Logging Off the Embedded Web Server
- 6. Gateway's ini File Configuration
- Secured ini File
- Modifying an ini File
- The ini File Content
- The ini File Structure
- The ini File Example
- Networking Parameters
- System Parameters
- Web and Telnet Parameters
- Security Parameters
- RADIUS Parameters
- SNMP Parameters
- SIP Configuration Parameters
- Voice Mail Parameters
- ISDN and CAS Interworking-Related Parameters
- Number Manipulation and Routing Parameters
- E1/T1 Configuration Parameters
- Channel Parameters
- Configuration Files Parameters
- 7. Using BootP / DHCP
- 8. Telephony Capabilities
- Working with Supplementary Services
- Configuring the DTMF Transport Types
- Fax & Modem Transport Modes
- Event Notification using X-Detect Header
- ThroughPacket™
- Dynamic Jitter Buffer Operation
- Configuring the Gateway’s Alternative Routing (based on Conn
- Call Detail Report
- Supported RADIUS Attributes
- Trunk to Trunk Routing Example
- Proxy or Registrar Registration Example
- SIP Call Flow Example
- SIP Authentication Example
- 9. Networking Capabilities
- 10. Advanced PSTN Configuration
- 11. Advanced System Capabilities
- 12. Special Applications
- 13. Security
- 14. Diagnostics
- 15. SNMP-Based Management
- SNMP Standards and Objects
- Carrier Grade Alarm System
- Cold Start Trap
- Third-Party Performance Monitoring Measurements
- TrunkPack-VoP Series Supported MIBs
- Traps
- SNMP Interface Details
- SNMP Manager Backward Compatibility
- Dual Module Interface
- SNMP NAT Traversal
- SNMP Administrative State Control
- AudioCodes’ Element Management System
- 16. Configuration Files
- Appendix A. Selected Technical Specifications
- Appendix B. Supplied SIP Software Kit
- Appendix C. SIP Compliance Tables
- Appendix D. The BootP/TFTP Configuration Utility
- Appendix E. RTP/RTCP Payload Types and Port Allocation
- Appendix F. RTP Control Protocol Extended Reports (RTCP-XR)
- Appendix G. Accessory Programs and Tools
- Appendix H. Release Reason Mapping
- Appendix I. SNMP Traps
- Appendix J. Installation and Configuration of Apache HTTP Server
- Appendix K. Regulatory Information
SIP User's Manual 1. Overview
Version 5.0 25 October 2006
Proxy and Registrar Authentication (handling 401 and 407 responses) using Basic or
Digest methods. Accepted challenges are kept for future requests to reduce the
network traffic.
Single gateway Registration or multiple Registration of all gateway endpoints.
Supported methods: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, INFO,
REFER, UPDATE, NOTIFY, PRACK and SUBSCRIBE.
Modifying connection parameters for an already established call (re-INVITE).
Working with a Redirect server and handling 3xx responses.
Early Media (supporting 183 Session Progress).
PRACK reliable provisional responses (RFC 3262).
Call Hold and Transfer Supplementary services using REFER, Refer-To, Referred-By,
Replaces and NOTIFY messages.
Supports RFC 3711, Secured RTP and Key Exchange according to <draft-ietf-
mmusic-sdescriptions-12>.
Supports RFC 3489, Simple Traversal of UDP Through NATs (STUN).
Supports RFC 3327, Adding ‘Path’ to Supported header.
Supports RFC 3581, Symmetric Response Routing.
Supports RFC 3326, Reason header.
Supports RFC 4028, Session Timers in SIP.
Locating SIP Servers (RFC 3263).
An Offer/Answer Model with Session Description Protocol (SDP) (RFC 3264).
Supports network asserted identity and privacy (RFC 3325 and RFC 3323).
Supports Tel URI (Uniform Resource Identifier) according to RFC 2806 bis.
Remote party ID <draft-ietf-sip-privacy-04.txt>.
Supports obtaining Proxy Domain Name(s) from DHCP (Dynamic Host Control
Protocol) according to RFC 3361.
RFC 2833 Relay for Dual Tone Multi Frequency (DTMF) digits, including payload type
negotiation.
DTMF out-of-band transfer using:
• INFO method <draft-choudhuri-sip-info-digit-00.txt>
• INFO method, compatible with Cisco gateways
• NOTIFY method <draft-mahy-sipping-signaled-digits-01.txt>.
SIP URI: sip:”phone number”@IP address (such as 1225556@10.1.2.4, where
“122556” is the phone number of the source or destination) or
sip:”phone_number”@”domain name”, such as 122556@myproxy.com. Note that the
SIP URI host name can be configured differently per called number.
Supports RFC 4040, RTP payload format for a 64 kbit/s transparent data.
Can negotiate coder from a list of given coders.
Responds to OPTIONS messages both outside a SIP dialog and in mid-call.
Generates SIP OPTIONS messages as Proxy keep-alive mechanism.
Representing trunk groups in tel/sip Uniform Resource Identifiers (URIs) according to
<draft-ietf-iptel-trunk-group-04>.
The number of total and free channels is published in a 200 OK response to an
OPTIONS request. The gateway uses the X-Resource header in the following format:
‘X-Resource: telchs=100/240;mediachs=0/0’,
Where ‘telchs’ specifies the number of free tel channels / total tel channels.
For more updated information on the gateway’s supported features, refer to the latest
Mediant & TP Series SIP Digital Gateways Release Notes.