SIP Servers

These days Linux Live CD VoIP Servers are available that can run directly from the CD.

SIP Servers


VoIP SIP Live CD Server

These days Linux Live CD VoIP Servers are available that can run directly from the CD. You don't have to even install software on your machine. The Linux LiveCD VoIP Server is very easy to use and it can be used to provide a Vonage type service. You can even use the CD to create a voip pbx for a campus or business with up to thousands of SIP phones.

This Live CD VoIP Server is based based on the Open Standard SIP Express Router (SER) and Asterisk. It can serve as a SIP Proxy, VoIP PBX, VoIP gateway or Class 5 Softswitch.

Features of the Live VoIP Server:

  • Easy Web user administration and real-time accounting.
  • All in one solution to VoIP and SIP enable your business.
  • Allows you to make your own SIP numbering plan. Centrex service.
  • Can be connected to multiple A-Z wholesale termination providers and to your own PSTN termination gateway/router.
  • Can do Least Cost Routing
  • Includes nat traversal, media server for conference call bridge, voicemail to email, incomming DID numbers.
  • Commercial pre-paid, post-paid and flat rate account support. No calling card (no b2bua in base system).
  • Requires no software installation - it is a liveCD.
  • Supports any SIP soft or hardware phones, such as popular XTen, Cisco ATA 186, Grandstream, Sipura, Bugetone, Linksys PAP2 and more.
  • Supports SIP for Video Conferencing (Xten / CounterPath EyeBeam)
  • Supports Ecrypted SIP with Xten / Counterpath Secure Xten Pro
  • Requires a PC with fixed ip connection to the internet. 128 MBytes of RAM, CDRom reader and ide hard disk to store call and user database and web site.
  • Remote ssh configuration and administration help

You can get the cd from SIP Express Router
SIP Express Router (ser) is a high-performance, configurable, free SIP ( RFC3261 ) server . It can act as SIP registrar, proxy or redirect server. SER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available.
Its performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and rapidly growing user population.
SER's configuration ability meets needs of a whole range of scenarios including small-office use, enterprise PBX replacements and carrier services.

SIP Server
The OnDO SIP Server, a SIP Registrar and Proxy Server, registers and authenticates users, and routes calls between user agents. The product has original NAT traversal (SIP NAT) functionality as well as flexible control routing functions. With OnDO SIP Server, you can use SIP hardphones, SIP softphones, and SIP-PSTN Gateways for VoIP communications. Version 1.5 now has Enhanced Dial Plan functionality.

With the growing popularity of SIP technology, and the variety of SIP devices to choose from, compatibility between SIP servers and SIP devices is fast becoming a concern for many users. Version 1.5 includes improved dial plan features that increase compatibility between SIP compliant products and services, and provide added capability for creating complex and powerful call control rules.

Solaris 10 OS As a SIP Server
You can use your Solaris 10 OS as SIP Server. This article explains how to set up a system running the Solaris 10 OS as a Session Initiation Protocol (SIP) server, with persistence via the MySQL database, which is bundled with the Solaris 10 release. The authors also show the setup of SIP-based clients to use SER on the Solaris 10 OS and demonstrate voice communications.

Bea Weblogic SIP Server
BEA WebLogic SIP Server is designed to operate as the core application container at the service control plane of the operator's network, providing SIP-based session control capabilities to next-generation, converged, IP communication services. It offers network operators, enterprises, service providers, and network equipment providers with a telecom industry-proven, converged application container platform on which to standardize the services layer of their next-generation networks. Session Initiation Protocol (SIP) is an application layer signaling protocol defined by the Internet Engineering Task Force (IETF) for initiating, modifying and terminating communication sessions between endpoints in an IP network. SIP is rapidly becoming the de facto underlying session control protocol that service providers and network operators are using as the foundation for SDP implementations.The convergence of SIP and J2EE technologies is a natural convergence of IT and communication technologies, and when deployed within the SDP, are enabling new IP-based services,such as VoIP.

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