siproxd-0.5.2 released
Thomas Ries <[email protected]> Sat, 31 Jan 2004 00:33:53 +0100 (CET)
| Newsgroups | gmane.network.siproxd |
|---|---|
| Message-ID | <[email protected]> |
Release Notes for siproxd-0.5.2
===============================
After some time, again a release of siproxd. This release
does not introduce new features, it contains mainly fixes
and enhancements to follow better RFC3261.
IPCHAINS & IPTABLES (netfilter) proxy support has been removed
as the RTP relay now seems to work reliable.
Major changes since 0.5.1:
- removed IPCHAINS & IPTABLES (netfilter) proxy support
- RTPPROXY correction: match RTP ports crosswise -
use one single port (and socket) on each side (inbound/
outbound) to send and receive RTP traffic for every
active stream (patch from Christof Meerwald).
- Handle with Route headers pointing to myself
- Include mandatory branch parameter in added via headers
- 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 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
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.
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.
-----
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0.5.2
=====
30-Jan-2004: - If RTP proxy is disabled, don't rewrite incomming
SDP bodies (patch from Robert Högberg)
29-Jan-2004: - new doc/RFC3261_compliance.txt and comments in the
code that refer to the RFC.
28-Jan-2004: - don't die on INVITE requests that include no Contact
header - which is legal. (patch from Robert Högberg)
- RTP proxy: don't try to forward empty RTP packets
- renamed some variables of rtp_proxytable_t to make
better sense (changed meaning in fullduplex RTP proxy)
27-Jan-2004: - added doc/KNOWN_BUGS
- better branch parameter calculation (via header),
now honors RFC3261 for stateless proxies (section 16.11)
- SIP request: remove a Route-header pointing to myself.
This was an issue with Linphone 0.12.1.
(patch from Robert Högberg).
- removed IPCHAINS & IPTABLES (netfilter) proxy support
- RTPPROXY correction: match RTP ports crosswise -
use one single port (and socket) on each side (inbound/
outbound) to send and receive RTP traffic for every
active stream (patch from Christof Meerwald).
22-Jan-2004: - ./configure option: --enable-static to build
a completely statically linked executable
- REGISTER honors the expires parameter
of the contact header
- Contact header of REGISTER response must be
rewritten back to the local (true) URL
18-Jan-2004: - security_check_raw:
size check: >= 16 bytes
- at exit, check registration file to be writable
- no WARNING if SIP user-agent header is not supplied.
- Call logging: distinguish between In & Out
- include branch parameter for via headers
--
GnuPG: pub 1024D/87BCDC94 2000-03-19 Thomas Ries <[email protected]>
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- 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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