148 lines
5.5 KiB
Plaintext
148 lines
5.5 KiB
Plaintext
Release Notes for siproxd-0.8.1
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===============================
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Major changes since 0.8.0:
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- new Plugins:
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plugin_prefix: add a prefix on outgoing calls
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plugin_regex: regular expression rewriting (To header) for outgoing calls
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- adjustable pthrad stack size (smaller memory footprint on small
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embedded systems like OpenWRT routers)
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- plus various bugfixes
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Upgrade Notes 0.8.0 to 0.8.1:
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- merge your configuration file siproxd.conf (new config options)
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General Overview:
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- SIP (RFC3261) Proxy for SIP based softphones hidden behind a
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masquerading firewall
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- basic support for SIP TCP transport
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- Support for PRACK messages (RFC3262)
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- Support for UPDATE messages (RFC3311)
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- SIP UDP and TCP supported
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- Works with "dial-up" conenctions (dynamic IP addresses)
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- Multiple local users/hosts can be masqueraded simultaneously
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- Access control (IP based) for incoming traffic
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- Proxy Authentication for registration of local clients (User Agents)
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with individual passwords for each user
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- May be used as pure outbound proxy (registration of local UAs
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to a 3rd party registrar)
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- runs on various operating systems (see below)
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- Full duplex RTP data stream proxy for *incoming* and *outgoing*
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audio data - no firewall masquerading entries needed
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- Port range to be used for RTP traffic is configurable
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(-> easy to set up apropriate firewall rules for RTP traffic)
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- RTP proxy can handle multiple RTP streams (eg. audio + video)
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within a single SIP session.
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- Symmetric RTP support
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- Symmetric SIP signalling support
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- Supports running in a chroot jail and changing user-ID after startup
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- All configuration done via one simple ascii configuration file
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- Logging to syslog in daemon mode
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- RPM package (Spec file)
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- The host part of UA registration entries can be masqueraded
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(mask_host, masked_host config items). Some Siemens SIP phones seem to
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need this 'feature'.
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- Provider specific outbound proxies can be configured
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- Can run "in front of" a NAT router.(in the local LAN segment)
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- supports "Short-Dials"
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- configurable RFC3581 (rport) support for sent SIP packets
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Requirements:
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- pthreads (Linux)
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- glibc2 / libc5 / uClibc
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- libosip2 (3.x.x)
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Mainly tested on:
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- CentOS 5, 32bit Linux
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This is the main development and testing environment. Other platforms
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are not extensively tested.
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Builds on (tested by dev-team or reported to build):
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- Linux: CentOS/RedHat EL
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( Fedora 64bit )*
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( WRT54g (133mhz mipsel router))*
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(- FreeBSD: FreeBSD 4.10-BETA )*
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(- OpenBSD: OpenBSD 3.4 GENERIC#18 )*
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(- SunOS: SunOS 5.9 )*
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(- Mac OS X: Darwin 6.8 )*
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* Note: As the compile farm of sourceforge.net has been discontinued our
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building test possibilities are now very limited. Currently
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no explicit testing for systems/distributions other than
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CentOS/RHEL (x86 architecture) is made. We'll be looking into
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possibilities to perform some broader testing in future.
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Of course, external testers are welcome :-)
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Reported interoperability with softphones:
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- Grandstream BudgeTone-100 series
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- Linphone (local and remote UA) (http://www.linphone.org)
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- Kphone (local and remote UA) (http://www.wirlab.net/kphone/)
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- MSN messenger 4.6 (remote and local UA)
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- X-Lite (Win XP Professional)
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- SJPhone softphone
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- Asterisk PBX (using a SIP Trunk, masqueraded via siproxd)
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- Ekiga
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- FreePBX
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Reported interoperability with SIP service providers:
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- Sipgate (http://www.sipgate.de)
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- Stanaphone (SIP Gateway to PSTN)
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- Sipcall.ch (Swiss VoIP provider)
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- Ekiga
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If you have siproxd successfully running with another SIP phone
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and/or service provider, please drop me a short note so I can update
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the list.
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Known interoperability issues with SIP service providers:
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- callcentric.com (afaik callcentric fails with "500 network failure"
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during REGISTER if more than one Via header is
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present in a SIP packet. Having multiple Via headers
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is completely in compliance with RFC3261. This might
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be related to their "NAT problem avoidance magic".
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There is nothing that can be done within siproxd
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to avoid this issue as callcentric does not comply
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with the SIP specification.
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- asterisk PBX Asterisk has an issue finding the proper peer
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if multiple peers originate from the same IP/port
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tuple (a is the case if multiple phones are proxied
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via siproxd to the same asterisk instance).
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This is caused by the SIP implementation in
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asterisk (chan_sip).
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Note: This seems to be no longer valid with
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asterisk version 1.6 and up.
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Known bugs:
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- SRV DNS records are not yet looked up, only A records
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There will be more for sure...
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If you port siproxd to a new platform or do other kinds of changes
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or bugfixes that might be of general interest, please drop me a
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line. Also if you intend to include siproxd into a software
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distribution I'd be happy to get a short notice.
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-----
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Signatures for siproxd-0.8.1.tar.gz archive:
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MD5 Hash: 1a6f9d13aeb2d650375c9a346ac6cbaf
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SHA-256 Hash: df2df04faf5bdb4980cbdfd5516a47898fc47ca1ebc2c628aa48305b20a09dad
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GnuPG signature:
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-----BEGIN PGP SIGNATURE-----
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Version: GnuPG v1.4.5 (GNU/Linux)
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iD8DBQBOGgSbB2xLpFxU+GURAt/gAJ9uWS01n7Tr7G7HlX8Zp8+W33OYZACfX69S
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mTpcbCWOxuoKDp5R3GWZ+zg=
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=BFqD
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-----END PGP SIGNATURE-----
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GnuPG: pub 1024D/87BCDC94 2000-03-19 Thomas Ries (tries at gmx.net)
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- Fingerprint = 13D1 19F5 77D0 4CEC 8D3F A24E 09FC C18A 87BC DC94
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- Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
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VoIP: sip:17476691342@proxy01.sipphone.com | sip:431783@fwd.pulver.com
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