SipX
From Wikipedia, the free encyclopedia
- The correct title of this article is sipX. The initial letter is shown capitalized due to technical restrictions.
| SIPfoundry sipX ECS (IP PBX) | |
 Screenshot of the sipX Configuration Manager |
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| Developer: | SIPfoundry |
|---|---|
| Latest release: | 3.6 / February 12, 2007 |
| OS: | Linux |
| Use: | IP telephony |
| License: | GNU Lesser General Public License |
| Website: | http://www.sipfoundry.org/ |
sipX ECS (Enterprise Communications Server) is an open source software implementation of a Session Initiation Protocol (SIP) based communications system (IP PBX). Similar to a traditional private branch exchange (PBX), it allows a number of attached telephones to make calls to one another, and to connect to other telephone services including the PSTN and SIP Trunking services. sipX is homed at SIPfoundry.
sipX ECS became an open source project early 2004 when Pingtel Corp contributed the codebase to the not-for-profit organization SIPfoundry. The name sipX insignifies a SIP switch or router and refers to the native SIP based call control system around which sipX is architected. In contrast to most other IP PBX systems that use SIP as a transport protocol but implement features and feature interactions in a proprietary way internal to the IP PBX, the sipX project implemented a native SIP based system with all features and all feature interactions implemented in SIP. This approach led to a modular and therefore highly scalable system.
The sipX ECS system serves as a reference implementation of the SIP standard. It is used at SIPIT interoperability events organized by the SIP Forum to test interoperability of SIP solutions from many different vendors. An automated SIP interoperability portal based on sipX is provided for free by Pingtel at http://interop.pingtel.com that is primarily used by SIP phone manufacturers for SIP compliance and advanced feature testing.
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[edit] Applications
sipX ECS is used as a Voice over IP communications system by enterprises ranging from about 200 users up to about 10,000 users. sipX replaces conventional private branch exchanges with a software based application that runs on standard server hardware and uses standard operating systems. The largest publicly announced deployment is at Amazon.com serving over 5,000 users (announced October 6, 2006).
[edit] Availability
sipX is available on multiple Linux distributions, including Red Hat Enterprise Linux, Fedora Core, CentOS and others.
Like most IP PBXs (open source or not), sipX provides the following traditional PBX features:
[edit] Features
[edit] PBX System Features
- Aliasing facility
- Automatic Route Selection
- Auto-restart services after power failure using watchdog facility
- Browser-based configuration system
- Call Admission Control
- Codec support (Enhanced G.711 A-law and μ-law)
- Dynamic call forwarding
- Hunt groups
- Message waiting indication
- Multi-site / multi-location station and gateway
- Multi-station appearance
- Outbound call blocking
- System Security
- URI mapping engine for call routing and inter-company (/domain) SIP calls
- Web services APIs for Config server (SOAP)
[edit] Voicemail Features
- wav file messages
- Browser-based interface
- DTMF interface
- Editable message headers
- Email notification of new voicemail messages
- Folders for message organization
- https-based message storage
- Multiple user customizable voicemail greetings
- Operator escape from anywhere
[edit] Auto Attendant Features
- Customizable auto attendant message for main greeting
- Dial by extension
- Dial by name
- Operator escape from anywhere
- Customizable IVR menus with VXML
[edit] User Features
- Browser-based user interface
- Call coverage
- Call forward
- Call hold / retrieve
- Call waiting / retrieve
- Calling line identification
- Calling party name identification
- Conferencing
- Direct inward dial (DID)
- Message waiting indication
- Multiple call appearance
- Multi-station appearance
[edit] SIP Implementation
- RFC 3261 Session Initiation Protocol using both UDP and TCP transports
- Advanced call control using RFCs
- 3515 Refer Method
- 3891 Referred-By header
- 3892 Replaces header
- Provide for consultative and blind transfer and third party call controls
- RFC 3263 Locating SIP Servers - use of DNS SRV records for call routing control and server redundancy.
- RFC 3581 Symmetric Response Routing (rport)
- RFC 3265 SIP Event Notification - for phone configuration and
- RFC 3842 Voice mail message waiting indication (MWI)
- RFC 3262 Reliable Provisional Responses
- RFC 2833 Out-of-band DTMF tones
- RFC 3264 Offer/Answer model for SDP for Codec Negotiation
- Early media (SDP in 180/183)
- Delayed SDP (SDP in ACK)
- Re-INVITE: Codec change, hold, off-hold
- Route/Record-Route header fields
- Configurable RTP/RTCP ports
- Configurable SIP ports
sipX supports the use of Ethernet-attached SIP phones. To attach ordinary (non-VoIP) phones or PSTN lines to the PBX, IP/PSTN gateways are used. sipX supports a number of commercially-available gateways.
sipX is distinguished from most other open source VoIP PBXs by several characteristics:
- All call signaling is handled using the SIP protocol natively (vs. gatewaying SIP to some other signaling protocol, e.g. as done in the Asterisk PBX).
- The sipX components handle call signaling, but once a call is set up, the voice (media) packets are sent directly between the endpoints involved. This allows most of the sipX components to be agnostic about the media and its encodings. E.g., SIP-based Videophones can communicate without increasing the load on the sipX components.
- The architecture of the system is non-monolithic; the sipX components (proxy, media server, etc.) communicate between each other via SIP and can be run on different hosts (or replaced with other SIP components), allowing high scalability, high reliability, and ease of integrating other SIP components.
- The system administrative interface is web-based (vs. a command-line interface).
sipX adheres to the SIP philosophy of implementing many features with significant support in the endpoints (telephones, gateways, voicemail systems) rather than entirely in the core components (proxy). This improves scalability but makes many features dependent on support in the endpoints of the telephone system.
[edit] Distributions
sipX is dual-licensed free software, released under the Lesser GNU General Public License (LGPL). A commercially packaged version is available from Pingtel,[1] similar to the Red Hat distribution of Linux or the Digium distribution of Asterisk.
[edit] Hard Phones
[edit] Plug & Play Managed:
Plug & Play in the SipX context refers to phones directly supported by its configuration system, sipXconfig, thus allowing configuration of the phone without the need to use the telephone keypad or the embedded webserver of the telephone. Getting a phone up and running with a default configuration is as simple as entering its MAC address into sipXconfig, assigning a user (line) and generating a profile. The telephone will then be configured by the sipXconfig application and the SipX server upon startup of the telephone device.
- Polycom SoundPoint & SoundStation Phones
- SNOM SIP Phones
- Grandstream SIP Phones
- Cisco SIP 79xx Phones
[edit] Manually Configured:
- ACT SIP Phone
- Sipura/Linksys SIP Phones
- ZyXEL Prestige WiFi SIP Phone
- Pingtel xpressa Phone
[edit] Softphones
- KPhone
- Xten X-Lite & eyeBeam Phones
- SNOM 360 Softphone
- Windows Messenger
- LinPhone
- SJPhone
- sipXphone
- Ekiga Phone (former GnomeMeeting)
- OpenWengo
