[Ekiga-devel-list] Fwd: [Asterisk-video] [Linphone-users] New softphone demo from antisip!
- From: "michel memeteau" <michel memeteau gmail com>
- To: "Ekiga development mailing list" <ekiga-devel-list gnome org>
- Subject: [Ekiga-devel-list] Fwd: [Asterisk-video] [Linphone-users] New softphone demo from antisip!
- Date: Tue, 15 Jan 2008 10:45:36 +0100
Hi all , just for you to know : Linphone devs have a new softphone on the road :
---------- Forwarded message ----------
From: Klaus Darilion <
klaus mailinglists pernau at>
Date: 15 janv. 2008 09:29
Subject: Re: [Asterisk-video] [Linphone-users] New softphone demo from antisip!
To: Aymeric Moizard <jack atosc org
>
Cc: Development discussion of video media support in Asterisk <asterisk-video lists digium com>, osip atosc org,
linphone-users nongnu org
Hi Aymeric and Asterisk users.
FYI: I've just tested the softphone with a Asterisk and a 3G video
call(nokia 6630): In direction 3G-->SIP the video is fine. In direction
SIP->3G* the video is there, but bad quality (block artefacts ...).
regards
klaus
*make sure to set upload bandwidth to 64kbit
Aymeric Moizard schrieb:
>
> Dear users,
>
> Most of your already know that I've started a company named antisip 3
> years ago. Since that time, I have improved a lot my osip2 and eXosip2
> projects and have put a lot of effort in contributing to oRTP & the
> mediastreamer2 projects.
>
> This helped me much for my commercial project named amsip which now
> offer a complete SIP sdk including presence, instant messaging, as well
> as a good set of audio & video codecs on all major platforms.
>
> Among capabilities, you can find in amsip/eXosip2/mediastreamer2:
> * amazing NAT traversal capabilities based on the ICE specification
> allowing many calls to be peer to peer even if both correspondant
> are behing a NAT!
> * video conferencing
> * performant H264 codec using intel primitives
> * bandwidth negotiation to adapt video framerate & compression.
>
> But text explanation are usually not enough! That's why I'm proud
> to announce the first version of my own softphone.
>
>
http://sip.antisip.com/download/emansip-setup/emansip-setup-v411-rc10.exe
>
> You have to create a new account on
sip.antisip.com if you don't
> already have any:
>
>
http://sip.antisip.com/
>
> Once you have created an account, you'll receive a mail where you have
> to confirm your account creation before you can use the service. If
> you don't receive the mail, please ask me and I'll confirm myself
> your account.
>
> Current features for this softphone:
> * calls
> * encryption (TLS & SRTP)
> * music on hold, mute, record converstation
> * presence
> * video calls (configure your upload/download bandwidth)
> * audio conference
> * version limited to speex/PCMU/PCMA/gsm
> * version limited to H263-1998 video codec
> * Try streaming files in conversation!!!
> * Try video conference (configure your upload bandwidth to minimum value!)
>
> I'm working on adding Instant Messaging: it will appear in a very
> few days.
>
> As it's the first official version of this softphone, it will certainly
> contains a few bugs. I hope you'll be kind enough to report them to me!
>
> It's time for me to wish you a happy new year and success for
> all your software developments,
>
> For any business, support related questions, you can call me at
> <
sip:antisip sip antisip com>
>
> tks to all,
> Aymeric MOIZARD / ANTISIP
> amsip -
http://www.antisip.com> osip2 -
http://www.osip.org
> eXosip2 -
http://savannah.nongnu.org/projects/exosip/>
>
>
> _______________________________________________
> Linphone-users mailing list
>
Linphone-users nongnu org>
http://lists.nongnu.org/mailman/listinfo/linphone-users
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