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Internet phone free long distance

Internet Phone System

Knowledge and Technical Guide for Administrators and Resellers

1.1.1 - Components of a SIP Network
1.1.2 - Provisioning Overview
1.1.3 - Security Overview
1.1.3.1 - Proxy Servers
1.1.4 - SIP Services
1.1.4.1 - Basic Services
1.1.4.2 - Enhanced Services
1.1.4.3 - PSTN Interworking
1.2 - Network Address Translation (NAT) Traversal
1.2.1 - What is a NAT or NAPT (Network Address Port Translator)?
1.2.2 - VoIP-NAT Interworking
1.3 - Voice Quality Overview
2 - Hardware Overview
2.1 - Phone Adapter LED Status
2.2 - Broadband Router (RT31P2) LED Status
3 - Software Configuration Mechanisms
3.1 - Configuration Profile Formats
3.1.1 - Using the Supplemental Profile Compiler
3.1.2 - Encrypting and Compressing XML configuration files
3.2 - Secure Initial Configuration
3.3 - Web Interface
3.3.1 - Web Interface Conventions
3.3.2 - Administration Privileges
3.3.3 - Basic and Advanced Views
3.4 - Functional Configuration URLs
3.4.1 - Upgrade URL
3.4.2 - Resync URL
3.4.3 - Reboot URL
3.5 - Configuration via the IVR (PAP2 only)
4 - Configuration Parameters
4.1 - Data Types
4.1.1 - Conventions
4.2 - Provisioning Related Parameters
4.2.1 - Firmware Upgrade
4.2.2 - Provisioning Server Redundancy
4.2.3 - Configuring the Web Server and IVR
System Configuration
4.3 - Basic Networking Configuration
Network Configuration
4.4 - Basic Account Configuration
4.5 - Configuration for NAT Traversal
4.6 - Media and SDP (Session Description Protocol) Configuration
4.6.1 - DTMF and Hookflash
4.6.2 - Codec and Audio Settings
4.6.3 - Dynamic Payload Types and SDP Codec Names
4.6.4 - Secure Media Implementation:
4.6.5 - Outbound Call Codec Selection Codes:
4.7 - Supplementary Services
4.7.1 - Supplementary Services activated internally
4.7.2 - Call Forwarding Implemented internally
4.7.3 - Supplementary Services implemented in the service provider network
4.8 - Dial Plan Configuration
4.8.1 - Speed Dialing Settings
4.9 - Progress Tone and Ring Configuration
4.9.1 - Distinctive Ring and Other Ring Settings
4.9.2 - Progress Tones
4.10 - Less Frequently Used Paramters
4.10.1 - Advanced Protocol Parameters
4.10.2 - Additional User Account Information
4.10.3 - Per-Line Polarity Settings
4.10.4 - Additional Timer Values (sec)
4.10.5 - Miscellaneous Parameters
5 - Expected Feature Behavior
5.1 - Originating a Phone Call
5.2 - Receiving a Phone Call
5.3 - Caller ID
5.4 - Calling Line Identification Presentation (CLIP)
5.5 - Calling Line Identification Restriction (CLIR) – Caller ID Blocking
5.6 - Call Waiting
5.7 - Disable or Cancel Call Waiting
5.8 - Call-Waiting with Caller ID
5.9 - Voice Mail
5.10 - Attendant Call Transfer
5.11 - Unattended or “Blind” Call Transfer
5.12 - Call Hold
5.13 - Three-Way Calling
5.14 - Three-Way Ad-Hoc Conference Calling
5.15 - Call Return
5.16 - Automatic Call Back
5.17 - Call FWD – Unconditional
5.18 - Call FWD – Busy
5.19 - Call FWD - No Answer
5.20 - Anonymous Call Blocking
5.21 - Distinctive / Priority Ringing and Call Waiting Tone
5.22 - Speed Calling – Up to Eight (8) Numbers or IP Addresses
6 - Troubleshooting
6.1 - Call Statistics Reporting
6.2 - Enabling Logging and Debugging
6.3 - Error and Log Reporting
6.4 - Internal Error Codes
6.5 - Provisioning and Upgrade result codes
6.6 - Table of SIP Response Codes (Error Codes)
7 - Summary of Implemented Features and Specifications
7.1 - Data Networking Features
7.1.1 - MAC Address (IEEE 802.3)
7.1.2 - IPv4 – Internet Protocol Version 4 (RFC 791) upgradeable to v6 (RFC 1883)
7.1.3 - ARP – Address Resolution Protocol
7.1.4 - DNS – A Record (RFC 1706), SRV Record (RFC 2782)
7.1.5 - DiffServ (RFC 2475) and ToS – Type of Service (RFC 791/1349)
7.1.6 - DHCP Client – Dynamic Host Configuration Protocol (RFC 2131)
7.1.7 - ICMP – Internet Control Message Protocol (RFC792)
7.1.8 - TCP – Transmission Control Protocol (RFC793)
7.1.9 - UDP – User Datagram Protocol (RFC768)
7.1.10 - RTP – Real Time Protocol (RFC 1889) (RFC 1890)
7.1.11 - RTCP – Real Time Control Protocol (RFC 1889)
7.2 - Voice Features
7.2.1 - SIPv2 – Session Initiation Protocol Version 2 (RFC 3261-3265)
7.2.1.1 - SIP Proxy Redundancy – Static or Dynamic via DNS SRV
7.2.1.2 - Re-registration with Primary SIP Proxy Server
7.2.1.3 - SIP Support in Network Address Translation Networks – NAT
7.2.2 - Codec Name Assignment
7.2.3 - Secure Calls
7.2.4 - Voice Algorithms:
7.2.4.1 - G.711 (A-law and mµ-law)
7.2.4.2 - G.726
7.2.4.3 - G.729A
7.2.4.4 - G.723.1
7.2.5 - Codec Selection
7.2.6 - Dynamic Payload
7.2.7 - Adjustable Audio Frames Per Packet
7.2.8 - Fax Tone Detection Pass-Through
7.2.9 - DTMF: In-band & Out-of-Band (RFC 2833) (SIP INFO *)
7.2.10 - Call Progress Tone Generation
7.2.11 - Call Progress Tone Pass Through
7.2.12 - Jitter Buffer – Dynamic (Adaptive)
7.2.13 - Full Duplex Audio
7.2.14 - Echo Cancellation – Up to 8 ms Echo Tail
7.2.15 - Voice Activity Detection with Silence Suppression & Comfort...
7.2.16 - Attenuation / Gain Adjustment
7.2.17 - Signaling Hook Flash Event
7.2.18 - Configurable Flash / Switch Hook Timer
7.2.19 - Configurable Dial Plan with Interdigit Timers
7.2.20 - Message Waiting Indicator Tones – MWI
7.2.21 - Polarity Control
7.2.22 - Calling Party Control – CPC
7.2.23 - International Caller ID Delivery
7.2.24 - Streaming Audio Server – SAS
7.2.25 - Music On Hold – MOH
7.3 - Security Features
7.3.1 - Multiple Administration Layers (Levels and Permissions)
7.3.2 - HTTP Digest – Encrypted Authentication via MD5 (RFC 1321)
7.3.3 - HTTPS with Client Certificate
7.4 - Administration and Maintenance Features
7.4.1 - Web Browser Administration and Configuration via Integral Web Server
7.4.2 - Telephone Key Pad Configuration with Interactive Voice Prompts
7.4.3 - Automated Provisioning & Upgrade via TFTP, HTTP and HTTPS
7.4.4 - Periodic Notification of Upgrade Availability via NOTIFY or HTTP
7.4.5 - Non-Intrusive, In-Service Upgrades
7.4.6 - Report Generation and Event Logging
7.4.7 - Syslog and Debug Server Records
8 - List of all configuration parameters
 

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