partysip-announce
[Top][All Lists]
Advanced

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

[Partysip-announce] libosip-0.8.8 and partysip-0.4.5 major release!


From: Aymeric Moizard
Subject: [Partysip-announce] libosip-0.8.8 and partysip-0.4.5 major release!
Date: Wed, 26 Jun 2002 13:36:33 +0200 (CEST)

I'm finally there!
I worked the past week on the windows port of partysip.
(the ppl/win32 replaces the ppl/unix library). It's now
running fine on *linux*, *OpenBSD*, and *windows*. (I need some
people to validate the solaris and osf ports, please test it
and send a report!)

About oSIP:

Working for the first time on windows, I have to warn user of
previous version of oSIP that they should get a lot of trouble!!
The contributions that I get on year ago is just plain wrong!
The method sthread_join(), instead of waiting for thread to
terminate just kill them even if they are unfinished!

I suppose that windows users experienced some troubles... :)
>From now on, I will try to maintain and validate myself the windows
port... :(

About Partysip:

Partysip has reached its first step of developpement:
* registrar capability.
* redirect/stateless/statefull support.
* record-routing mode is available and optionnal.

Interoperability issues:
partysip is (try) to be compliant with the latest draft-09.
But almost all HW phones (or even software ones) are not. This fact
makes it difficult to successfully handle calls when record-routing
mode is enabled! Even if partysip has the backward compatibility
mechanism with rfc2543, softwares and HW phones are not exactly
compliant with it! If you are not testing partysip: DO NOT
enable record-routing.

I've been able to use 2 UA (josua) and 2 proxy (partysip) and
I tested call flows with and without record-routing and it's
working fine! I'm using the "draft-ietf-sipping-call-flows-00.txt"
to verfiy the validity of the call flows. (but this spec contains
a few errors I'll report...)

Configuration :
 plenty of users are trying to use partysip with the original
configuration file (partysip.conf) which is defined to be used for
local test on 127.0.0.1. partysip can't support using two different
interface and if partysip is running on this interface, it won't
be able to handle UA that are not defined to run on 127.0.0.1.
Remember to edit at least the serverip entry in the config file
to be the IP of your host for the network you are testing.

Next step for partysip:
* clean the code (especially make sfp.c and slp.c more readable :)
* call plugins evrywhere they must be called (this is tested
  only for new incomnig request, new outgoing request in uap,
  but not tested for requests and responses for stateless and
  statefull mode as no operations are actually needed.
* new plugins:
  * accounting (with record-route enabled)
  * filtering with a 'static' table for forwarding/rejecting request

josua, partysip and osip pre release are available on
osip.atosc.org/download. Please report any success or failure tests!
The new version is running on anode.atosc.org port 5060.

Hope this is enough for you.
Enjoy
Aymeric




reply via email to

[Prev in Thread] Current Thread [Next in Thread]