similar to: Typo in vorbis RTP spec

hi kuhlen, what you said is correct. i am talking about how you are going to arrange these codewords into an array, i.e. in the function _make_decode_table. there he uses node bytes and leaf words for memory management. i got a 24 bit platform. so if i assume that max. codeword length that could be possible as 24 bits can i allocate a memory of (2 * used entries - 2), to arrange the whole tree in

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 >>>

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

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, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

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

Hello List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP UA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP address which is declared in the Connection Information (c) in the SDP, obviously reaching nowhere. I need RTP to be sent to the

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a

Today I faced a problem. Please help to solve this problem. Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel

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

Hello List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP UA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP address which is declared in the Connection Information (c) in the SDP, obviously reaching nowhere. I need RTP to be sent to the

Hello ! A question to all 'Wheel-reinventers': I can build the huffmantrees by hand (on paper) but how to code it? Are there any good URLs out there? Or does the spec supply sufficient information? I tried figure out how oggdec (debugging) does this, but I couldn't get the clou. Thank you Dominik --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project

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

Hello everybody! Some questions concerning oggenc: 1 why is it called oggenc? (vorbisenc would make more sense) 2 Is it true, that oggenc uses only some predefined codebooks (depending on -q)? 2.1 Are they just "random" books, or were they "optimized" in any way 2.2 Is it possible, that some codebooks are stored in the header, but are never used (even in long (>3min) files)?

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

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

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

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)