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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.
-----
Signatures for siproxd-0.8.1.tar.gz archive:
MD5 Hash:
SHA-256 Hash:
GnuPG signature:
GnuPG: pub 1024D/87BCDC94 2000-03-19 Thomas Ries (tries at gmx.net)
- 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@proxy01.sipphone.com | sip:431783@fwd.pulver.com
|
