similar to: Stun & codec

* Brethren, It's a sad day in our community. Please join me in a moment of silence for the death of responsible journalism. Silence.....................good enough. This article goes on to tell about Pingtel's announcement of forming the "first open source community aimed at creating SIP based servers".

The H.264 codec patent by Qualcomm has been ruled invalid by a San Diego Federal jury: http://www.eetimes.com/news/semi/showArticle.jhtml?articleID=197001066 . That means that H.264 codecs can now be written, distributed and revised freely under any license their authors choose, including GPL, public domain, or any other, and $free now that royalties are no longer required. How does H.264

Here's t link: http://www.informationweek.com/story/showArticle.jhtml;jsessionid=JUEFVG ENEA01YQSNDBCCKH0CJUMEKJVN?articleID=165702588 The bottom line is that they compare retail VOIP providers like Comcast Cable, Time-Warner Cable, AT&T, Vonage, Packet8 et al. Their methodology seems sound. Their conclusion is that retail VOIP services don't yet match the PSTN for reliability &

Any thoughts on articles like this one - http://www.theregister.co.uk/2008/02/21/microsoft_goes_open/. This mentions the Vista API, but I wonder about earlier API's. I also saw something the other day about source code for .Net being released as a "learning tool". Of course, all of this is in a bid to get Office XML approved as an ISO standard. They don't just give stuff away

The Skype network is circulating a virus that has appeared there before: http://www.informationweek.com/news/showArticle.jhtml?articleID=198500135 . The virus sends a URL to other Skype users in the infected user's contacts, which the target Skype displays as clickable. Clicking downloads the virus. Asterisk supports features like these, in combination with certain clients (which aren't

Hey, just came across this article. http://www.eetimes.com/news/latest/showArticle.jhtml;jsessionid=ZLCZZJMVGJYNYQSNDLSCKHA?articleID=201202250 . Many of you would be already aware of this. It mentions flac also. i feel it is a good segment(stereo wireless headsets over bluetooth) to push flac given very light decoder and using encoder at low compression(to reduce complexity). Though very less is

Hope this isn't too far OT, but its relevant to us. From isn.attrition.org >http://www.informationweek.com/story/showArticle.jhtml?articleID=20300851 > >By W. David Gardner >TechWeb News >May 13, 2004 > >As voice over IP sweeps across the high-tech landscape, many IT >managers are being lulled into a dangerous complacency because they >look upon Internet phoning

Hi, I am trying to register SIP clients which are behind NAT on different network. In order to achieve this goal I think I need STUN Server . I downloaded STUN Server from http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz But I don't know how to install/configure it. And please advice me that STUN server is good idea for this scenario? Thanks in advance Wazb

I have been trying out several stun servers with Asterisk 1.6.0.9 and 1.6.1.0 and I keep getting the following message: [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]: chan_sip.c:2695 ast_sip_ouraddrfor: stun failed [May 26 12:26:35] WARNING[16174]:

HI all! I'm looking for some infos to configure stun server support for a SIP peer. I've installed Asterisk 1.4.3, but searching for stun support in chan_sip (sip.conf) i've found nothing, only a "misterious" externip = stun... But where i have to put the ip of stun server? No infos around Google and forum! :-) Thank all, regards -- Marco Ciacci Asterisk Admin Windows

Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040715/b37cee4b/attachment.htm -------------- next part -------------- ? Hi friends I have some doubt in connecting my firefly3rd party softphone from windows machine to asterisk server in linux . My asterisk is behind the "Port Restricted NAT". I am using STUN server to cross the

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: >>[snip] >Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk >can handled the NAT traversal all by itself with Qualify (as John points >out) disabling the NOTIFY will not change anything. > >The NOTIFY will in no way affect the status - unreachable/reachable. > >Another problem with the SIPURA is

I have an asterisk 13 server behind NAT on a dynamic IP Address. It is running the PJSIP Stack It is registering to another asterisk 13 server that is on a Static IP outside of the firewall at a different location (also on the PJSIP Stack). How do we implement STUN/ICE on the server behind the dynamic Address. It does not appear to be registering properly without knowing the NAT pubic

I'm finishing up deploying an Asterisk (Trixbox) box at work. Wow, I thought Asterisk was cool by itself, but Trixbox has made just about everything turnkey. Great stuff! So... we're using Grandstream GXP-2000 handsets to connect to the Trixbox, which sits on our DMZ with a single public IP. I need the phones to work from random places behind NAT, as well as in the office. I'm using

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

Not sure if it's alright to talk about this here??? compiled the STUN server from Vovida on RedHat 7.3. Looks simple to configure. It isn't starting...it tries to for a long time and then just craps out. Here is my config:/etc/sysconfig/stund #!/bin/echo Not to execute. # Path to stund STUND=/usr/sbin/stund # Set the required args for STUND STUNDPRIMARYHOSTNAME=208.x.x.x # The hostname

Hello I was wondering if someone knew of an application that could check that the user has a firewall and a broadband connection that will work OK with Asterisk and VoIP. The app would first perform some bandwith + jitter tests, and will then call a STUN server to check that the firewall isn't symetric. BTW, is Asterisk now STUN-capable, or is it still to map ports manually on the firewall