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RELNOTES
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Release Notes for siproxd-0.8.4
===============================
Major changes since 0.8.3:
-
New plugins:
-
Upgrade Notes 0.8.3 to 0.8.4:
- Merge the configuration file
General Overview:
- SIP (RFC3261) Proxy for SIP based softphones hidden behind a
masquerading firewall
- plugin system allows loading extensions to accomplish various tasks
- Support for PRACK messages (RFC3262)
- Support for UPDATE messages (RFC3311)
- SIP UDP and TCP supported
- Works with "dial-up" connections (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 appropriate 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 signaling 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
Plugins:
- plugin_fix_fbox_anoncall
This plugin attempts to work-around some SIP issues with
Fritzbox devices and anonymous calls. Fritzbox devices do change their
Contact header when answering an anonymous call (suppressed CLID) - this
in turn confuses siproxd. This plugin attempts to work around this by
sanitizing the Contact Header before processing.
- plugin_siptrunk
Plugin to handle SIP Trunks where using *one* single SIP account a
whole number block is routed. Please read the comments in the config
file section.
- plugin_codecfilter
Allows blacklisting of codecs and removes those from any passing SDP
payload in both (incoming and outgoing) directions. This allows the
proxy to force the exclusion of particular codecs in the negotiation
between a local UA and a remote side.
- plugin_stripheader
Allows to strip particular headers from SIP messages. Useful if your
provider chokes on some headers included by your local UA.
- plugin_regex
Applies an extended regular expression to the 'To' URI.
- plugin_prefix
Unconditionally prefixes all outgoing calls with a prefix.
- plugin_stun
Uses an external STUN server to determine the public IP
address of siproxd. Useful for "in-front-of-NAT-router"
scenarios.
- plugin_fix_DTAG
This plugin attempts to work-around some SIP issues with
T-ONLINE SIP (as of 2015). T-Online.de sends broken Via headers in
responses, causing the received SIP response to be discarded by
any SIP client that properly checks the Via chain.
- plugin_fix_bogus_via
Incoming (from public network) SIP messages are checked for broken
SIP Via headers. If the IP address in the latest Via Header is
part of the list below, it will be replaced by the IP where the
SIP message has been received from.
- plugin_shortdial
Quick Dial (Short Dial)
Ability to define quick dial numbers that can be accessed by
dialing e.g. "*01" from a local phone.
Requirements:
- pthreads (Linux)
- glibc2 / libc5 / uClibc
- libosip2 (4.x.x)
Mainly tested on:
- CentOS
This is the main development and testing environment. Other platforms
are not extensively tested. However, the code should be quite portable
and build on many UNIX/Linux flavors.
Builds on (tested by dev-team or reported to build):
- Linux: CentOS/RedHat
- FreeBSD: FreeBSD 10
Reported interoperability with softphones:
- SNOM series
- Fritzbox UAs
- 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
- SJPhone softphone
- Asterisk PBX (using a SIP Trunk, masqueraded via siproxd, chan_sip driver)
- Ekiga
- FreePBX
- Yealink series
Reported interoperability with SIP service providers:
- Sipgate (http://www.sipgate.de)
- Stanaphone (SIP Gateway to PSTN)
- Sipcall.ch (Swiss VoIP provider)
- Nexphone.ch (Swiss VoIP provider - via resellers)
- Ekiga
- DTAG (Deutsche Telecom AG) -> requires plugin_fix_DTAG to work around
some issues with this provider
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.
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.
-----
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