- rel 0.7.2

This commit is contained in:
Thomas Ries 2009-10-11 10:22:33 +00:00
parent 3a50376f3e
commit efc88e5b6c
2 changed files with 26 additions and 7 deletions

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@ -1,5 +1,6 @@
0.7.2
=====
11-Oct-2009: - Released 0.7.2
30-Jul-2009: - RTP proxy could get confused and mess up RTP streams
if two local UAs did use the same usernames but register
on different providers. A call from UA1 to UA2 then could

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@ -2,10 +2,10 @@ Release Notes for siproxd-0.7.2
===============================
Major changes since 0.7.1:
-
- mostly bugfixes related to aborting RTP streams
Upgrade Notes 0.7.1 to 0.7.2:
- none
General Overview:
- SIP (RFC3261) Proxy for SIP based softphones hidden behind a
@ -49,10 +49,11 @@ Requirements:
Mainly tested on:
- Fedora 32bit Linux
This is my main development and testing environment. Other platforms
are not extensively tested by myself.
are not extensively tested.
Builds on (tested by dev-team or reported to build):
- Linux: Fedora
CentOS
( Fedora 64bit )*
( WRT54g (133mhz mipsel router))*
(- FreeBSD: FreeBSD 4.10-BETA )*
@ -63,9 +64,9 @@ Builds on (tested by dev-team or reported to build):
* 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
Fedora (x86 architecture) are made. We'll be looking into
Fedora/CentOS (x86 architecture) is made. We'll be looking into
possibilities to perform some broader testing in future.
Of course, external help will be heartly welcome :-)
Of course, external help will be welcome :-)
Reported interoperability with softphones:
- Grandstream BudgeTone-100 series
@ -76,6 +77,7 @@ Reported interoperability with softphones:
- SJPhone softphone
- Asterisk PBX (using a SIP Trunk, masqueraded via siproxd)
- Ekiga
- FreePBX
Reported interoperability with SIP service providers:
- Sipphone (http://www.sipphone.com)
@ -86,6 +88,7 @@ Reported interoperability with SIP service providers:
- Ekiga
- Gizmo (actually sipphone.com)
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.
@ -100,6 +103,14 @@ Known interoperability issues with SIP service providers:
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).
Known bugs:
- SRV DNS records are not yet looked up, only A records
There will be more for sure...
@ -112,10 +123,17 @@ distribution I'd be happy to get a short notice.
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GnuPG: pub 1024D/87BCDC94 2000-03-19 Thomas Ries (tries at gmx.net)