similar to: Updated Speex RTP Internet Draft

The latest revision of the draft RTP Profile is attached for review. This will be submitted to the IETF Audio-Video Transport Working Group for consideration immediately, so if you have any more comments, let us know. In addition, we will be applying for an official MIME type. Note that the AVP code and the MIME type in this latest revision have been changed from "SPX" to

On Tue, Dec 17, 2002 at 11:55:21PM -0800, Greg Herlein wrote: > If such a thing happens, discussion of the RTP profile draft > would be most welcome - please get responses back to the > list! Now, if this were finalised before the conference then we could do a demo and use it for broadcasting the lectures streams around the world... What is currently the best way of doing this? I'm

Hi all We are about to send an updated version of the internet draft "RTP Payload Format for the Speex Codec" to the IETF AVT working group. Before submitting we would like your input, if you have any comments or input please send them to the mailing list. If we don't get any comments in 1 week (by 22. May 2007) we will go ahead and submit it to the IETF. Of course you can comment

Looks good to me. Jean-Marc Alfred E. Heggestad a ?crit : > Hi > > Please find an updated version of the Speex I-D attached. The only > change is addition of the copyright conditions in Appendix A, > as requested by Ivo. > > Many thanks for your input. > > I will give you a few more days before submitting to AVT working group > > > /alfred > > Ivo

Do not forget to add the "Copying conditions" to the RFC. Check http://wiki.debian.org/NonFreeIETFDocuments That page contains a section titled "Template for RFC authors to release additional rights". To follow that guideline a section like the following should be added: x. Copying conditions The author(s) agree to grant third parties the irrevocable right to

Hello, What's the purpose of the 'sr' sdp parameter ? The sample rate is already given in the a=rtpmap line ? Simon Le dim 29/06/2003 à 12:12, philkerr@elec.gla.ac.uk a écrit : > Hi all, > > Please find below an updated Speex Internet Draft document. > > It would be good if we could book some time for discussion on Speex at the IETF > meeting in Vienna (scheduled

Hi all, Please find below the -01 update to draft-herlein-speex-rtp-profile, as submitted to the IETF. Regards Phil <p>-------------------8<-----------------------------------8<--------------------- <p><p>Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin

> Otherwise Greg, can you send the latest version of the RTP draft so I > can put it on the site (the current one is getting old)? Attached for all to see. Greg -------------- next part -------------- Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin draft-herlein-speex-rtp-profile-06

All: Attached please find yet another RTP profile revision (v5). You can also find the document at: http://www.herlein.com/downloads/speex/docs/ Changes: - added vbr, cng, ebw, sr optional parameters to MIME - added vbr, cng, ebw a=fmtp options for SDP use - added required document attributes for submission to IETF and IANA (format and author contact information). Note that we

The latest revision of the draft profile for use of Speex in RTP is attached. We plan on submitting this - or a modified version of this, based on immediate feedback - to the IETF on Monday for consideration at the next meeting. Major differences in this revision are: - removed the discussion in the MIME section. It's a duplicate of the SDP discussion anyway, and may or may not match the

>>>>> "Greg" == Greg Herlein <gherlein@herlein.com> writes: Greg> The IETF has already assigned -02: ... to indicate it's Greg> expired. I'll ask if they want to use -02 or -03. They posted a policy statement a couple of weeks back. The expired notice is to be replaced by the next version of the draft if any is forthcoming. So it'll be -02.

Hi, We'd like to announce the first draft for the Speex RTP profile. It was written essentially by Greg Herlein, with some help from Simon Morlat and I. We'd like to get some feedback on it before it is sent to the IETF. Basically this will allow all SIP based VoIP applications using Speex to inter-operate. For those interested, there's already Simon's LinPhone (www.linphone.org)

Alfred E. Heggestad wrote: > <...> > > If we don't get any comments in 1 week (by 22. May 2007) we will go ahead > and submit it to the IETF. Of course you can comment on it also after it > has been submitted, but we would like to get the input from the Speex > community first.. > thanks for all the input. please find attached an updated version of the draft. I

Hi Please find an updated version of the Speex I-D attached. The only change is addition of the copyright conditions in Appendix A, as requested by Ivo. Many thanks for your input. I will give you a few more days before submitting to AVT working group /alfred Ivo Emanuel Gon?alves wrote: > Do not forget to add the "Copying conditions" to the RFC. > > Check

Looks good. Two things though: - I think the draft should be names -02 since the last one was -01 (I'm pretty sure). - The header in the current document should be changed (they say -01) Jean-Marc Le lun 01/03/2004 à 20:17, Greg Herlein a écrit : > > > > Is this the latest draft? > > > >

Hi, We are trying to use Speex for voice encoding along with network streaming in our live-streaming software. We have had little luck trying to find players which can support demuxing of RTP payload for speex (draft-herlein-speex-rtp-profile-02.txt) . Can someone please let me know if any such players are available which have support for this draft? Thanks in advance! -Kiran --------------

This portion of the RFC is gramatically incorrect and confusing: The RTP payload MUST be padded to provide an integer number of octets as the payload length. These padding bits MUST be all zero. This padding is only required for the last frame in the packet, and only to ensure the packet contents ends on an octet boundary. <p> -- Ben Greear <greearb@candelatech.com>

On Wed, Jan 14, 2004 at 07:35:48PM -0500, Jean-Marc Valin wrote: > > Does the stable release support the new draft? > > Yes. Version 1.0.1 makes it easier (than 1.0.0) to implement the draft, > but even with 1.0.0 it's not that hard. The only thing that changed is > the "terminator" that allows a variable number of frames per packets > (instead of a negotiated

> Page 3: > > To be compliant with this specification, implementations MUST support > 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. > The sampling rate MUST be 8, 16 or 32 kHz. > > There is a type above after (narrowband), there is a " extra character. > > I don't understand what is the motivation to specify "SHOULD

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing