About Repro

From reSIProcate
Revision as of 16:08, 4 May 2005 by Rjsparks (talk | contribs) (Key Features)

Jump to: navigation, search

Overview

What is repro?

repro is an open-source, free SIP server.

repro provides SIP proxy, registrar, redirect, and identity services. These services are the foundation needed to run a SIP service.

Where can repro be used?

  • As the central rendezvous service for peer-to-peer voice, IM, and presence services
  • As the core of large scale internet telephony services
  • As a tool to enforce policy at the boundary between networks or domains

Key Features

  • Correct and comprehensive implementation of the relevant standards from
 the SIP working groups
  • Support for multiple transport protocols over both IPv4 and IPv6
  • Rigorous security mechanisms, including the newest SIP Security IETF efforts
  • Simple user management through an embedded configuration web server
  • Use of readily available databases (currently Berkeley DB) to store user data.
  • Extendable to support provider enhanced features while processing requests

Upcoming Features

  • MySQL or Postgres storage of user data
  • Acting as a Presence Server

Support

Several companies are in the process of putting together commercial support plans for repro, reSIProcate and DUM that are targeted at application developers and service providers. More detail to follow.

How to Participate

Project Details

Current Features

  • Transports: UDP, TCP, TLS (v4 and v6)
  • Platforms: Win32, Linux, Mac OS X
  • RFC 3261 compliant proxy and registrar
  • RFC 3263 compliant: NAPTR, SRV, A, AAAA
  • Extendable features
  • In-memory location server
  • Embedded web server and user database for basic administration
  • Full support for draft-ietf-sip-identity-04
  • IPv6

Roadmap

Goals for 0.1 release (beginning of May)

  • source tar ball available
  • binaries available for Windows (exe), Linux (rpm), MacOSX (dmg)
  • allow web admin to modify and delete users and static routes
  • add access lists for devices that don't need to authenticate (like gateways)
  • tested according to test plan

Goals for 0.2 release

  • high availability (distribution of user database and registrations across a geographically diverse server farm)
  • call forwarding
  • certificate/credential service (draft-ietf-sipping-certs-01.txt)

Goals for 0.3 release

  • support for the GRUU extension and outbound-only connections

Test Plan

Basic Calling Scenarios

  • REGISTER/401 REGISTER/200 un-REGISTER/200
  • INVITE/180 200/ACK BYE/200 both directions
  • INVITE/180 CANCEL/200/487/ACK both directions
  • INVITE/180 4xx/ACK as caller
  • Make / receive basic call behind a NAT (with UDP)

Devices

  • Sipura SPA-2000
  • Cisco 7960
  • Xten eyeBeam
  • snom 190
  • Cisco 3600 gateway
  • SER proxy (running at iptel.org)
  • Grandstream Bugetone
  • sipXphone
  • Vegastream phone
  • Jasomi B2BUA
  • M5T proxy server

Scenario tests

  • Make a call with TCP
  • Make a call with TLS
  • Forking test
  • Spiral test
  • Loop test
  • Basic load test - create 1000 test users, register them at certain rate, make basic calls at certain rate, specify average call duration before sending BYE.