Re: [Partysip-dev] libosip-0.8.8 and partysip-0.4.5 major release!

[Tim Na]

Great job!
About Solaris, we recently upgade it to Solaris 8 and partysip is workin fine 
with
the new CVS version.
I have been a bit busy with implementing a simple plugin for IM service.

So far so good :-)
If I see a problem, I will report it.
oh BTW, gethostbyname_r() detection is not still working.  I will try to
look into it later. :-)
Thanks.

Regards,
Tim Na

Samsung Electornics.
Mobile Telecommunication Department.
----- Original Message ----- 
From: "Aymeric Moizard" <address@hidden>
To: <address@hidden>; <address@hidden>; <address@hidden>
Sent: Wednesday, June 26, 2002 8:36 PM
Subject: [Partysip-dev] libosip-0.8.8 and partysip-0.4.5 major release!


> 
> 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
> 
> 
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