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Re: [Linphone-users] New softphone demo from antisip!


From: Klaus Darilion
Subject: Re: [Linphone-users] New softphone demo from antisip!
Date: Tue, 15 Jan 2008 09:29:25 +0100
User-agent: Thunderbird 2.0.0.9 (Windows/20071031)

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:address@hidden>

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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