similar to: Speex is the default codec for Jabber's Jingle VoIP

Ivo Emanuel Gon?alves wrote: > Just a heads-up, I received confirmation that Speex is now the default > codec for the Jabber's Jingle VoIP protocol. Which we hope to finalize soon for broader adoption. :) > While not the default in Google's Jabber, Speex has been reported to > work on Google Talk as well as of last year. BTW, my contacts on the Google Talk team report that

Peter Saint-Andre a ?crit : > Ivo Emanuel Gon?alves wrote: >> Just a heads-up, I received confirmation that Speex is now the default >> codec for the Jabber's Jingle VoIP protocol. > > Which we hope to finalize soon for broader adoption. :) That's good to hear. Are you supporting wideband or just narrowband? Jean-Marc

Today, Adobe Systems Incorporated announced[1] that they will extend their Flash platform to support the proprietary H.264 and HE-AAC formats. The Xiph.Org Foundation has previously contacted Adobe to consider extending Flash to support the Open Media formats Vorbis, Theora, and Speex. Adobe developers replied by stating it would be unwise as it would increase the file size of the Flash Player,

Today, Adobe Systems Incorporated announced[1] that they will extend their Flash platform to support the proprietary H.264 and HE-AAC formats. The Xiph.Org Foundation has previously contacted Adobe to consider extending Flash to support the Open Media formats Vorbis, Theora, and Speex. Adobe developers replied by stating it would be unwise as it would increase the file size of the Flash Player,

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN

Hi! I just tried to call a friend using jingle, but I got refused. Errorcode was 502, he tried to call me, heard it ringing once and then it stopped. I used: originate jingle/gtalk_account/friend at jabber.linuxlovers.at [application] I'm registered to googletalk, but this should mean no harm, or should it. Once I was able to receive a text-message from him, but couldn't

Hello Jean: Big mistake!! One of the three things that a supplier should never discuss is politics ... that's business 101. Speex may be a very good product, but we will never use it because of the politics that you espouse ... some of us out here believe strongly in the free market. And we also know that monopolies can't exist because entrepreneurs will always exploit the

On May 5, 2008, at 6:58 PM, Jean-Marc Valin wrote: > Joshua Gargus wrote: >> Our app needs to support multiple media types, so GStreamer would >> seem >> to be the obvious choice. However, voice quality is paramount for >> us. Are there any disadvantages to using the Speex plugin for >> GStreamer as opposed to directly coding to the Speex API? > >

On Mon, Jun 04, 2007 at 03:20:05PM -0600, Peter Saint-Andre wrote: > >>Hi Luca, > >> > >>Thank you for working on the Theora and Vorbis drafts in the IETF's AVT > >>WG. Do you also plan to work on the Speex draft? > > > >other people are working on the speex I-D, Ralph should be able to tell > >you which to contact ^^; > > Hi Ralph,

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Peter Saint-Andre wrote: > On Mon, Jun 04, 2007 at 03:58:17PM -0700, Ralph Giles wrote: >> On Mon, Jun 04, 2007 at 03:20:05PM -0600, Peter Saint-Andre wrote: >> >>>>> Hi Luca, >>>>> >>>>> Thank you for working on the Theora and Vorbis drafts in the IETF's AVT >>>>> WG. Do you also plan to work on the Speex draft?

Hi all, All the examples I've come across seem to suggest configuring jabber.conf/jingle.conf/gtalk.conf for a real Google account. What about the scenario where the Asterisk server should connect to an account on a private Jabber server and using Jingle (voice calling over Jabber)? e.g. for the domain widgets.com: - there is a copy of ejabberd running on the same box as Asterisk, and

Hi Farhan, What is the purpose of the protocol? Just to be a binary form a SIP? I mean conferencing, PTT, chat we have in h.323, sip, jingle, iax? What is the reason for a new one? The NAT traversal has been fixed for a while for all those protocols also. Diana farhan@phonestack.com wrote: > Dear All, > > I have finished documenting my Lightweight Telephony Protocol. It is > now

Hi, we would like to build IM-Voice community for our students around Asterisk, Jingle, Jabber. Can we already test those features ? Anyone already running such setup? Any more info ? Thanks in advance, regards, Rob.

We interrupt your regularly scheduled program to bring you the following issue. RFC3534bis is on hold. I have addressed all feedback from Mr Westerlund, except for the following: From: Magnus Westerlund <magnus.westerlund[at]ericsson[dot]com> Date: Fri, 11 Apr 2008 13:55:28 +0200 Subject: Re: Updated 3534bis (was Re: Request for review of Ogg Media Types: video/ogg, audio/ogg,

Hi Jean-Marc, Alfred and Greg, Are you receiving the mails from IETF about draft-ietf-avt-rtp-speex The mails are not coming from AVT mailing list, but I think we are all 3 part of a minimal list (draft-ietf-avt-rtp-speex at tools.ietf.org) dedicated to latest discussion about the draft. I have answered some questions, but there are small changes and adaptation still required to the ietf

Dear all seems impossible to enter in the pages ofthe icecast product can you advice them? i start from here http://dir.xiph.org/ regards --------------------------------- Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3 -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm patching oggenc with this fix. If Conrad reports success in his test, I'll go ahead and patch all other implementations I can find. -Ivo

I'm patching oggenc with this fix. If Conrad reports success in his test, I'll go ahead and patch all other implementations I can find. -Ivo