Displaying 20 results from an estimated 40000 matches similar to: "Updated Speex RTP Internet Draft"
2004 Aug 06
1
RTP Profile Revision
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
2004 Aug 06
5
linux.conf.au and streaming (was Re: patch for libspeex)
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
2007 May 15
4
draft-ietf-avt-rtp-speex-01.txt
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
2007 Jun 07
1
draft-ietf-avt-rtp-speex-01.txt
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
2007 May 30
5
draft-ietf-avt-rtp-speex-01.txt
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
2004 Aug 06
0
Updated Speex RTP Internet Draft
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
2004 Aug 06
0
draft-herlein-speex-rtp-profile-01
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
2004 Aug 06
1
linux.conf.au and streaming (was Re: patch for libspeex)
> 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
2004 Aug 06
0
RTP Profile Revision v5
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
2004 Aug 06
0
Comments on New RTP Profile Document
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
2004 Aug 06
3
Re: Speex-RTP RFC questions
>>>>> "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.
2004 Aug 06
0
First draft for Speex RTP profile - Please send your comments
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)
2007 May 29
0
draft-ietf-avt-rtp-speex-01.txt
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
2007 Jun 07
0
draft-ietf-avt-rtp-speex-01.txt
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
2004 Aug 06
2
Speex-RTP RFC questions
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?
> > > >
2006 Sep 01
1
Players having RTP payload support for Speex
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
--------------
2004 Aug 06
4
Speex-RTP RFC questions
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>
2004 Aug 06
2
Speex-RTP RFC questions
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
2007 May 16
2
draft-ietf-avt-rtp-speex-01.txt
> 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
2004 Jan 26
3
X-Lite & Asterisk: Speex & iLBC not working?
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