230c3d3ffb
found by dmalloc memory poisoning - fix: new branch id hash calculation was not working properly (dont check return status in osip_xxx_to_str() functions - the docu does not say anything about them (looks like 0 == success) but better check the returnet string pointer value -> NULL = failed) This resulted in a memory leak an imporper calculated MD5 hash.
103 lines
3.8 KiB
Plaintext
103 lines
3.8 KiB
Plaintext
Release Notes for siproxd-0.5.2
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===============================
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Just before my vacations I'll make another release. Some new features
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have been added and some changes are still going on.
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After giving some thoughts, I decided to discontinue the IPCHAINS
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and IPTABLES support. Reasons for this are, the RTP relay does
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fulfill the same task (with the cost of more CPU time needed),
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but it is much easier to support (portability and support
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on other platforms) and has no need for any special firewall
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masquerading configuration.
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Major changes since 0.5.0:
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- Feature: Full duplex RTP proxy - masquerading for outgoing
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RTP stream is no longer needed.Only the RTP relay has been
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tested yet. However, I'm planning to discontinue IPCHAINS and
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IPTABLES support in the near future for the sake of less
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dependency of the OS and portability.
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- Feature: The BudgeTone SIP phones now DO work with the RTP relay.
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- Feature: UA registrations now survive a restart of siproxd
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(ongoing sessions however do not survive).
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- Fix: cope with changing RTP port numbers suring a session
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(e.g. kphone does it when doing HOLD/unHOLD)
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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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- 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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- Fli4l OPT_SIP (still experimental) available, check
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http://home.arcor.de/jsffm/fli4l/
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- supports Linux and FreeBSD (other BSD derivates not yet tested)
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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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- 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
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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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Requirements:
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- pthreads (Linux)
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- glibc2 / libc5 / uClibc
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- libosip2 (libosip1 is no longer supported!)
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Currently tested on:
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- Redhat 6.0 (Kernel 2.2.x, Glibc)
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- Redhat 7.2 (Kernel 2.4.x, Glibc)
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- SUSE 5.3 (kernel 2.0.x, libc5)
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- Redhat 7.2 build against uClibc
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- should run on others Linux distributions as well.
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- FreeBSD 4.7-STABLE (compilation)
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Reported to build on:
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- OpenBSD 2.9
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- Solaris2
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Reported interoperability (tested 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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If you can confirm other SIP phones working, please drop me
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a short note.
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Known bugs:
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There will be...
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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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md5sum for siproxd-0.5.2.tar.gz:
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GnuPG signature for siproxd-0.5.2.tar.gz archive:
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GnuPG: pub 1024D/87BCDC94 2000-03-19 Thomas Ries <tries@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:thomas@ries.homeip.net
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