similar to: Update on Vorbis RTP I-D

Hi All, Please find below an updated Vorbis-RTP Internet Draft document for review and discussion at the Xiph IRC meeting on Saturday. The changes in this version have been: Codebook caching mechanism Expanded SDP parameters Expanded MIME section Expanded introduction Packet loss section Minor tweaks and clarity changes to text There are probably some minor tweaks to the formatting needed

Hi all, Apologies in advance, this email is quite long. I've prepared an updated Vorbis RTP Internet Draft, which is a continuation of draft-moffitt-vorbis-rtp-00.txt which can be found below. If this new draft gets the ok I'd like to submit this to the AVT WG later this week. There are a number of changes over the original I-D, notably the changing of the M bit function in the RTP

Hi all, Below is the Vorbis RTP Internet Draft as sent to the AVT working group of the IETF. Comments and feedback is still welcomed from the Vorbis community. Cheers Phil ---------------------------8<-----------------8<------------------------ Network Working Group Phil Kerr Internet-Draft The Ogg Vorbis January 07, 2003 Community / OpenDrama

alpha. (that about sums it up) Will encode and play back via an sdp file and multicast on one computer (over the net if you transfer the sdp file over by hand). rc/rtenc3 and src/rtdec3 are in the style of encoder/decoder_example ...so this means you MUST edit it them to suit your system. For example, I specify my rtenc3 to multicast just on eth1 (to avoid pissing off my cable supplier on

Hi everybody, this is an update on the developed Ogg IETF standard documents and their status. All of these documents are in the process of discussion and have not yet been accepted as standards. <p>The following Internet-Drafts (I-Ds) have been prepared for standardisation and submitted to the IETF: 1) an I-D requesting to register "application/ogg" as a mime-type written by

Hi everybody, this is an update on the developed Ogg IETF standard documents and their status. All of these documents are in the process of discussion and have not yet been accepted as standards. <p>The following Internet-Drafts (I-Ds) have been prepared for standardisation and submitted to the IETF: 1) an I-D requesting to register "application/ogg" as a mime-type written by

Hi everybody, this is an update on the developed Ogg IETF standard documents and their status. All of these documents are in the process of discussion and have not yet been accepted as standards. <p>The following Internet-Drafts (I-Ds) have been prepared for standardisation and submitted to the IETF: 1) an I-D requesting to register "application/ogg" as a mime-type written by

http://www.xiph.org/ogg/vorbis/doc/draft-moffitt-vorbis-rtp-00.txt This is the Internet-Draft I'll be submitting tomorrow and hopefully presenting at the March IETF meeting. If you see anything major, let me know right away, I'll be submitting this in the morning. jack. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To

facing problem with originating webrtc calls 1-when iam doing call from webrtc iget ice working <--- SIP read from WS:91.196.158.205:1466 ---> INVITE sip:0669197533 at 77.91.132.9 SIP/2.0 Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315 Max-Forwards: 69 To: <sip:0669197533 at 77.91.132.9> From: "Anton" <sip:1065 at 77.91.132.9>;tag=5i21qaop43 Call-ID:

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

Hi again, after the discussion the last few days, I started doing some coding and first of all have a first very simple, quick and dirty demo client: http://www.j-ogg.de/rtp/index.html During the coding, I've run into a few problems: - I was planning to replace the identification header with appropriate entries in the SDP descriptor. However, the frame sizes are defined here (why not in

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone

Hi. I got bored of writing papers last week so I decided to write an app for sending and receiving multicast vorbis streams over RTP. A first version is available here: http://www.cdt.luth.se/~rolle/vorbis/ (I use the JOrbis decoder, so you need Java 2 with Javasound, e.g. JDK1.3. Since the decoding is done in Java, you probably need at least 300-350 MHz. Works with the IBM JDK1.3 on Linux on a

Hi. I got bored of writing papers last week so I decided to write an app for sending and receiving multicast vorbis streams over RTP. A first version is available here: http://www.cdt.luth.se/~rolle/vorbis/ (I use the JOrbis decoder, so you need Java 2 with Javasound, e.g. JDK1.3. Since the decoding is done in Java, you probably need at least 300-350 MHz. Works with the IBM JDK1.3 on Linux on a

The Asterisk Development Team would like to announce the release of Certified Asterisk 13.18-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.18-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

Hi, I am facing a (for me) strange problem. When placing a SIP-Call I normally get connected and the dialplan is executed. The Call-Flow is controlled by a PHP-Agi-Script. The script answers the call (via AGI-Command) and a simple voicefile is played. SOMETIMES(!) I get disconnected immediately after the Answer - without any reason. This happens about all fifth call. Later on you will find

The Asterisk Development Team would like to announce the release of Asterisk 15.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN

The Asterisk Development Team would like to announce the release of Asterisk 13.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.18.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: