[Siproxd-users] siproxd-0.5.1 released

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

Release Notes for siproxd-0.5.1
===============================

Just before my vacations I'll make another release. Some new features
have been added and some changes are still going on.
After giving some thoughts, I decided to discontinue the IPCHAINS
and IPTABLES support. Reasons for this are, the RTP relay does
fulfill the same task (with the cost of more CPU time needed),
but it is much easier to support (portability and support
on other platforms) and has no need for any special firewall
masquerading configuration.


Major changes since 0.5.0:
 - Feature: Full duplex RTP proxy - masquerading for outgoing
   RTP stream is no longer needed.Only the RTP relay has been
   tested yet. However, I'm planning to discontinue IPCHAINS and 
   IPTABLES support in the near future for the sake of less
   dependency of the OS and portability.

 - Feature: The BudgeTone SIP phones now DO work with the RTP relay.

 - Feature: UA registrations now survive a restart of siproxd
   (ongoing sessions however do not survive).

 - Fix: cope with changing RTP port numbers suring a session
   (e.g. kphone does it when doing HOLD/unHOLD)


General Overview:
 - SIP (RFC3261) Proxy for SIP based softphones hidden behind a
   masquerading firewall
 - 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)
 - Fli4l OPT_SIP (still experimental) available, check
   http://home.arcor.de/jsffm/fli4l/
 - supports Linux and FreeBSD (other BSD derivates not yet tested)
 - 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.
 - 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
 - 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'.

Requirements:
 - pthreads (Linux)
 - glibc2 / libc5 / uClibc
 - libosip2 (libosip1 is no longer supported!)

Currently tested on:
- Redhat 6.0 (Kernel 2.2.x, Glibc)
- Redhat 7.2 (Kernel 2.4.x, Glibc)
- SUSE 5.3 (kernel 2.0.x, libc5)
- Redhat 7.2 build against uClibc
- should run on others Linux distributions as well.
- FreeBSD 4.7-STABLE (compilation)

Reported to build on:
- OpenBSD 2.9
- Solaris2

Reported interoperability (tested 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)
 If you can confirm other SIP phones working, please drop me
 a short note.

Known bugs:
   There will be...

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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