similar to: rtp implementation

Hi! What happened to this document? Was it just left dying in the IETF system? Tor --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages sent to the

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, I'm wondering if the plans for streaming support include on demand streams via rtsp -- not just a multicast of one stream like a radio station (icecast)?

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

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, I am not sure if this is the right mailinglist (since I have an ogg problem not a vorbis)..... I am looking for some pointers on how to packetize an ogg stream. My idea is to use udp packets for transport, so I would need a way that would minimalize the impact of lost packets. Are there any code examples out there for this? Thanks, Jeroen p.s. Please cc me, I am not subscribed to this

Hi, I am not sure if this is the right mailinglist (since I have an ogg problem not a vorbis)..... I am looking for some pointers on how to packetize an ogg stream. My idea is to use udp packets for transport, so I would need a way that would minimalize the impact of lost packets. Are there any code examples out there for this? Thanks, Jeroen p.s. Please cc me, I am not subscribed to this

I'm trying to adapt decoder_example to to support more of an event driven model. I'm building an RTSP client/server where the client will have a function called everytime a new packet comes in. So, I want to queue up these packets to be played. My current attack is as follows (note this isn't very robust and I realized that): For first and second packet i grab all the header stuff

Is anyone working on Vorbis over RTP? It's been silent on this frontier for a while. I have been hearing from representatives of the IETF that they were not impressed by the previous draft, and that they were not coherent with RTP thinking and other RTP transport specs for various content. There exist a textbook on RTP which was released recently, if someone wants to learn it from the ground

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

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

Hi all, I'm in the final stages of putting a new Vorbis RTP draft together, the new sections are below. There are a number of open questions: 1.) At present there is a 16 bit length field for the codebooks. As they are spec'd as being unbound in length, but typically around 15K, are there situations where they may be greater than 64KB? The size limit can be extended to 2MB as there

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, I'm building an rtsp client/server that will support vorbis and I'm trying to determine the best way to play the vorbis content as the array is continuously being filled. Currently, I can download an ogg file via rtsp and write it out to a FILE and then play_file (from ogg123) it when it's done -- so the file is in the correct format. What would you recommend? -- Wayne Davis

How can i determine how many bits per sample are in .ogg files? Thanks. -- Wayne Davis - wfdavis@seas.upenn.edu - PGP Key Available --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject

I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:xxxx) line, and when I dial that extension I get: -- Called

Hi Folks, I have a situation where my Colo insists on a particular IP setup for my * server box. They allocate two blocks of IPs to my colo server. One set as my own (ex 20.20.20.20.4/30 - 4 ips) and the other as a transit lan (es 10.10.10.0/29). These are all public IP addresses and there is no NAT involved in. So essentially I have to set-up IP aliases in my Linux box as follows; Example:

Hi, I'm trying to get Theora working with a video conferencing application we are developing. This application sends the audio content via RTP over UPD, and as one would expect, there is a draft for how a Theora stream is supposed to be wrapped into an RTP stream. http://svn.xiph.org/trunk/theora/doc/draft-ietf-avt-rtp-theora-00.txt The problem is that it is not obvious to me how the raw

(hello all, this is my first writing. so please bear with me if I'm wrong anywhere.) orry to break too lately, but how is the RTP payload submission is going? could we see the new payload at March IETF? I agree that it would be fairy straightforward to make an RTP payload for ogg vorbis, assuming raw packets, AFAIK. using physical bitstream is, in this case, not adequate by the reasons in