Displaying 20 results from an estimated 30000 matches similar to: "Comments on New RTP Profile Document"
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
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
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
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
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
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
3
Updated Speex RTP Internet Draft
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 for 14th July). The cutoff for submission is
9:00am EDT, (GMT -04:00), 30th June.
Comments and feedback welcomed!
Regards
Phil
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 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 May 15
0
draft-ietf-avt-rtp-speex-01.txt
Here my comments:
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 support 8
kbps
2007 May 16
0
draft-ietf-avt-rtp-speex-01.txt
comment inline.
On Wed, 16 May 2007, Jean-Marc Valin wrote:
>> 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
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 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
2007 May 16
0
draft-ietf-avt-rtp-speex-01.txt
On Wed, 16 May 2007, Jean-Marc Valin wrote:
>>> The main idea is that Speex supports many bit-rates, but for one reason
>>> or another, some modes may be left out in implementations (e.g. for RAM
>>> or network reasons). What we're saying here is that you should make an
>>> effoft to at least support (and offer) the 8 kbps mode to maximise
>>>
2004 Aug 06
1
RTP Payload Format for the Speex Codec
Hi,
I have a few questions/remarks about the draft, in particular about the
SDP usage of Speex (section 5):
- Which samplerate should be used if you receive something like
m=3Daudio 8008 RTP/AVP 97
a=3Drtpmap:97 speex/8000
a=3Dfmtp:97 sr=3D16000;ebw=3Dultra
- There is an inconsistency in the description of the 'penh'parameter.
According to the description,
2004 Aug 06
0
Re: [AVT] Speex: Apologies for Missed Meeting (fwd)
All:
My presentation to the IETF on the proposed Speex payload format
did not occur. Details below.
Progress continues, though, towards getting the Speex payload
format approved.
Greg
---------- Forwarded message ----------
Date: Fri, 21 Mar 2003 08:58:23 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: Greg Herlein <gherlein@herlein.com>
Cc: avt@ietf.org
Subject: Re: [AVT]
2004 Aug 06
0
I-D ACTION:draft-herlein-avt-rtp-speex-00.txt (fwd)
All:
The latest draft RTP Payload Format for Speex is available via
the IETF. See below for details.
Greg
---------- Forwarded message ----------
Date: Tue, 09 Mar 2004 15:56:23 -0500
From: Internet-Drafts@ietf.org
To: IETF-Announce: ;
Subject: I-D ACTION:draft-herlein-avt-rtp-speex-00.txt
A New Internet-Draft is available from the on-line Internet-Drafts directories.
<p>
2014 Dec 11
0
PJSIP configuration question
I am not sure what you mean by the ful SIP signaling?
Here is the trace for the sip.conf which works successfully.
Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK
---- SIP ---
<--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 --->
INVITE sip:8005555555 at 64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP