[Asterisk-video] [Linphone-users] New softphone demo from antisip!

[Klaus Darilion]

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 at 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/
> 
> 
> 
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