siproxd-0.5.3 released

Release Notes for siproxd-0.5.3
===============================

Major changes since 0.5.2:
 - Support for connections between 2 local (masqueraded) UAs
 - Symmetric SIP signalling
 - handling of Max-Forwards header
 - A lot of smaller bugfixes

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, FreeBSD and Solaris
 - 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

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

Reported to build on:
- FreeBSD 4.7-STABLE
- 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.


-----
md5sum for siproxd-0.5.3.tar.gz: 	00a3d5d8d847da0db6cb9b16751f027a

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GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <[email protected]>
-      Fingerprint = 13D1 19F5 77D0 4CEC 8D3F  A24E 09FC C18A 87BC DC94
-      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
VoIP:  sip:17476691342-rne+wJe3e/u3Jhga2bR/9VaTQe2KTcn/@public.gmane.org | sip:[email protected]





0.5.3
=====
  14-Feb-2004:  - Released 0.5.3
                - Removed superfluous backslashes for line continuation
  11-Feb-2004:  - Use same SIP port number for RX & TX (-> support
                  symmetric SIP signalling)
   7-Feb-2004:  - Fix for local-UA to local-UA RTP proxying, symmetric
                  RTP was not working.
                - logging routines now use a MUTEX to be thread safe.
                - RTP proxy: fixed a bug that could lead to a deadlock
                  on very rapid HOLD/unHOLD sequences.
   1-Feb-2004:  - Added handling of Max-Forwards header
                - a detected via loop results in an 482 Loop detected
  31-Jan-2004:  - Allow 2 of my vias in header to let 2 UA's sitting
                  behind the same siproxd have conversation together
                  UA1 -->--\   >   /-->--\
                            siproxd       Registrar
                  UA2 --<--/   <   \--<--/
                - Redone code for evaluation if a received packet
                  is coming from the inbound or outbound network
                - RTP streams are now identified by call_id AND
                  USERNAME of the contact header. This provides
                  support for RTP proxying between 2 UAs sitting on the
                  inbound network. -> Calls between local UAs going via
                  siproxd should now work.
                  UA1 -->--\
                            siproxd 
                  UA2 --<--/ 
                - Rewriting of SUBSCRIBE messages should now work.
                - Removed obsolete prototypes from rtpproxy.h
                - If the RTP stream in one direction is found to be
                  aborted (sendto() failure), also stop the stream
                  for the opposite direction



-- 
GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <[email protected]>
-      Fingerprint = 13D1 19F5 77D0 4CEC 8D3F  A24E 09FC C18A 87BC DC94
-      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
VoIP:  sip:17476691342-rne+wJe3e/u3Jhga2bR/9VaTQe2KTcn/@public.gmane.org | sip:[email protected]

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