similar to: RTP Payload Format for the Speex Codec

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

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

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

> 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

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

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

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

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

=20 =20 12:53:41.358166 IP (tos 0xb8, ttl 127, id 0, offset 0, flags [none], = proto: UDP (17), length: 856) 189.8.113.170.5060 > 189.8.126.177.5060: SIP, = length: 828 INVITE sip:7002 at 189.8.126.177:5060;user=3Dphone SIP/2.0 Via: SIP/2.0/UDP 189.8.113.170:5060;branch=3Dz9hG4bKba4h2m2070fhnc4q20k1.1 Call-ID: d6dc25017b171144f35fb9e1c9c393a3 at 10.0.0.10

I am trying to make predictions with se's using a nlme (kew11.nlme below). I get an error indicating levels for a factor are not allowed. I have searched and read Rnews, MEMSS, MASS, R-Help, and other lists in Spanish where I found questions similar to mine but not solution. I do not really care about the method used. Any suggestions to obtain predictions with se's from an nlme

> 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

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

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

I have a search form on my rails site. once the a user has entered a query it displays the results in a table with an extra field button so it can be added to there own portfolio called add to my stocks. How do i get the data for the stock they want to add and put this into a new table called mystocks. Is using a form submit button the right choice? Any examples much appreciated Regards Nick

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)

Jean-Marc and others, I'm meeting informally with the AVT WG on monday, and formally presenting our latest drafts on friday. I've been following the vorbis/theora draft efforts, but haven't paid much attention to the speex work. Can you brief me on what I should talk about, what we'd like feedback on, what's new and different from other drafts as far as the speex work goes?

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

I have difficulties to understand how to use Silk GNC. I have enabled GNC (and VAD) something like this: param.setting.vad = 1; param.setting.cng = 1; param.setting.plc = 1; param.setting.penh = 1; status = pjmedia_codec_opus_set_default_param(&config, &param); I used debugger and printf to see that code executes in Opus CNG.c silk_CNG function noise generation loop. But still I cannot

qemu-stubdom was stripping the prefix from the "params" xenstore key in xenstore_parse_domain_config, which was then saved stripped in a variable. In xenstore_process_event we compare the "param" from xenstore (not stripped) with the stripped "param" saved in the variable, which leads to a medium change (even if there isn't any), since we are comparing something

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