[Siproxd-users] New Release 0.8.1

[Thomas R. <tr...@gm...>]

A new release of siproxd is available.

Release Notes for siproxd-0.8.1
===============================

Major changes since 0.8.0:
 - new Plugins:
   plugin_prefix: add a prefix on outgoing calls
   plugin_regex: regular expression rewriting (To header) for outgoing calls
 - adjustable pthrad stack size (smaller memory footprint on small
   embedded systems like OpenWRT routers)
 - plus various bugfixes

Upgrade Notes 0.8.0 to 0.8.1:
 - merge your configuration file siproxd.conf (new config options)

General Overview:
 - SIP (RFC3261) Proxy for SIP based softphones hidden behind a
   masquerading firewall
 - basic support for SIP TCP transport
 - Support for PRACK messages (RFC3262)
 - Support for UPDATE messages (RFC3311)
 - SIP UDP and TCP supported
 - Works with "dial-up" conenctions (dynamic IP addresses)
 - Multiple local users/hosts can be masqueraded simultaneously
 - Access control (IP based) for incoming traffic
 - Proxy Authentication for registration of local clients (User Agents)
   with individual passwords for each user
 - May be used as pure outbound proxy (registration of local UAs
   to a 3rd party registrar)
 - runs on various operating systems (see below)
 - Full duplex RTP data stream proxy for *incoming* and *outgoing*
   audio data - no firewall masquerading entries needed
 - Port range to be used for RTP traffic is configurable
   (-> easy to set up apropriate firewall rules for RTP traffic)
 - RTP proxy can handle multiple RTP streams (eg. audio + video)
   within a single SIP session.
 - Symmetric RTP support
 - Symmetric SIP signalling support
 - Supports running in a chroot jail and changing user-ID after startup
 - All configuration done via one simple ascii configuration file
 - Logging to syslog in daemon mode
 - RPM package (Spec file)
 - The host part of UA registration entries can be masqueraded
   (mask_host, masked_host config items). Some Siemens SIP phones seem to
   need this 'feature'.
 - Provider specific outbound proxies can be configured
 - Can run "in front of" a NAT router.(in the local LAN segment)
 - supports "Short-Dials"
 - configurable RFC3581 (rport) support for sent SIP packets

Requirements:
 - pthreads (Linux)
 - glibc2 / libc5 / uClibc
 - libosip2 (3.x.x)

Mainly tested on:
- CentOS 5, 32bit Linux
  This is the main development and testing environment. Other platforms
  are not extensively tested.

Builds on (tested by dev-team or reported to build):
- Linux:	CentOS/RedHat EL
(		Fedora 64bit		)*
(		WRT54g (133mhz mipsel router))*
(- FreeBSD:	FreeBSD 4.10-BETA	)*
(- OpenBSD:	OpenBSD 3.4 GENERIC#18	)*
(- SunOS:	SunOS 5.9		)*
(- Mac OS X:	Darwin 6.8		)*

* Note: As the compile farm of sourceforge.net has been discontinued our
        building test possibilities are now very limited. Currently
        no explicit testing for systems/distributions other than
        CentOS/RHEL (x86 architecture) is made. We'll be looking into
        possibilities to perform some broader testing in future.
        Of course, external testers are welcome :-)

Reported interoperability with softphones:
 - Grandstream BudgeTone-100 series
 - Linphone (local and remote UA) (http://www.linphone.org)
 - Kphone (local and remote UA) (http://www.wirlab.net/kphone/)
 - MSN messenger 4.6 (remote and local UA)
 - X-Lite (Win XP Professional)
 - SJPhone softphone
 - Asterisk PBX (using a SIP Trunk, masqueraded via siproxd)
 - Ekiga
 - FreePBX

Reported interoperability with SIP service providers:
 - Sipgate	(http://www.sipgate.de)
 - Stanaphone	(SIP Gateway to PSTN)
 - Sipcall.ch	(Swiss VoIP provider)
 - Ekiga


 If you have siproxd successfully running with another SIP phone
 and/or service provider, please drop me a short note so I can update
 the list.

Known interoperability issues with SIP service providers:
 - callcentric.com	(afaik callcentric fails with "500 network failure"
 			during REGISTER if more than one Via header is
			present in a SIP packet. Having multiple Via headers
			is completely in compliance with RFC3261. This might
			be related to their "NAT problem avoidance magic".
			There is nothing that can be done within siproxd
			to avoid this issue as callcentric does not comply
			with the SIP specification.

 - asterisk PBX		Asterisk has an issue finding the proper peer
			if multiple peers originate from the same IP/port
			tuple (a is the case if multiple phones are proxied
			via siproxd to the same asterisk instance).
			This is caused by the SIP implementation in
			asterisk (chan_sip).
			Note: This seems to be no longer valid with
			      asterisk version 1.6 and up.


Known bugs:
   - SRV DNS records are not yet looked up, only A records
   There will be more for sure...

If you port siproxd to a new platform or do other kinds of changes
or bugfixes that might be of general interest, please drop me a
line. Also if you intend to include siproxd into a software
distribution I'd be happy to get a short notice.


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