similar to: Peer-to-peer audio codec

Hi, Our webradio (Let's Go Zik - http://www.letsgozik.com) works with donation and partenship. I think it's the only way to keep a webradio alive for the moment... We are making our radio in a associative way... It's quite hard to "find" listeners. Currently we are nearly broadcasting for 60 simultaneous listeners (and approx. 5000 differents listeners per months)...

Hi, I'm not anywhere near an expert, but I had successfully used Darkice, TwoLame, and Icast231 to netcast a mp2 stream. If it will help, here is a snip from the related area of my darkice.cfg: [icecast2-1] format = mp2 bitrateMode = cbr bitrate = 384 quality = 1.0 server = 127.0.0.1 port = 32710 password = (duh!) sampleRate =

Hello! I have a quite complicated setup. In my network on each interface there is bandwidth limitation for each user. Booth outgoing (on interface itself) and incoming (attached IMQ) traffic. There is main HTB class which limits bandwidth for whole interface and HTB subclasses for each user. Filtering is done with hashing filters. This setup was working correctly. But now in the network I

Hi Oddsock, Like Clement, I am sure Nullsoft is still "offering" AOL's bandwidth since I think Nullsoft is not part of AOL anymore. About the new broadcasting methods, is the multicast technology already available? I have heard only few providers are equipped with multicast enabled routers. What about p2p streaming, is it really reliable? When I see Peercast's statistics, only

Hi all, I have read the larc howto and I need to apply a traffic shaper with this configuration: router / two interface etho and eth1 lan lan is on eth1 and on eth0 I''ve the dsl connection (1.2 Mbit / 256 kbit) I need to limit the bandwith towards lan and I''ve thought at HTB and tcng. I write the script belove. I want limit the p2p and ftp (ssh and irc) connection at 15

Thanks, The only issue here is that for each service I need to create 200 child classes if I have 200 clients... Let me explain the problem better I have the following connection from my ISP: (1024/1024) (rate/ceil) 1) First, I want to divide the 1024 into smaller pieces based on priority: 256/256 - P2P (I want to limit the P2P traffic as much as possible) 256/1024 - HTTP 256/1024 - FTP

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module

Hi, I''m new at traffic control and was reading up on HTB and using it to put an upper limit on traffic. I have a 256k DSL with 64k upload (which translates to about 5/6KB uploads). The machine running the P2P applications keeps filling up the 64K so my browsing from other machines in the network ends up being very slow. Since there are several P2P applications, I wanted to set the

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

Hello, after spending several hours reading archives, I decided to write new post. I successfully set up packet classification, made some basic HTB setup, made some simple graphical representation from HTB statistics data... BUT, I cannot figure out how to refine HTB to get this behaviour: I need that class "p2p" should be the last one to get some link capacity. If I set both RATE

Hi, I would like to make a QoS configuration on a linux based dsl router. It is for a server, so I want to shape outgoing traffic, incoming traffic should not be a problem as long as I have a quite assymetric connection. I would like to achieve the following goals: 1) To have one class (p2p) having all the available bandwith if there is no activity on other classes. 2) If another class (ftp

Hi all, i am trying to setup my first t1 in asterisk, i have been using asterisk for several years but ahve never needed a t1 line before. I have a sangoma card already in the server with 4fxo ports. Now i ordered two single port t1 line cards from sangoma for the two servers i am connecting with the point 2 point t1. I am currently at the location trying to set things up but have some

Hi all! I''m trying to shape traffic in a dorm''s network (4 mbit symmetrical internet link, about 200 computers, heavy p2p usage). The router is a p4xeon running linux 2.6.9 with the qnet patches (http://kem.p.lodz.pl/~peter/qnet/). When I activate ip_forward I get >20% packet loss and a lot of duplicates. Any ideas? I attach my shaping script. Thank you very much in advance,

Hi, I'm a 3rd year university student in the UK and for my dissertation (Final Year Project) I'm trying to create a peer to peer streaming network, where a audio stream gets sent around the network in a p2p fashion hopefully reducing the load from the source. I've decided to use ogg vorbis for my stream, and I have successfully created a small app that can read data from a .ogg file

oddcast DSP is what you want, it supports live input (called Advanced Recording) and does support streaming to Peercast...It currently does not support Vorbis metadata for live recording (since metadata for vorbis streaming is inserted into the ogg stream, as opposed to how it's done for mp3 in which the metadata is updated on the server via URL calls)... anyway, it's the closest thing to

From: Yonatan Maman <Ymaman at Nvidia.com> This patch series aims to enable Peer-to-Peer (P2P) DMA access in GPU-centric applications that utilize RDMA and private device pages. This enhancement is crucial for minimizing data transfer overhead by allowing the GPU to directly expose device private page data to devices such as NICs, eliminating the need to traverse system RAM, which is the

Hello everyone, I want an opinion from people who tryed different matching modules to match diferent types of traffic, especially p2p ones. I would like to hear which scales better as CPU usage and latency : ipp2p, iptables-p2p or l7-filter with the p2p patterns. I want to use one of them to block most of p2p (except maybe dc++ and emule which i want to shape). I would use the matching rules in

Hi, I have a setup like this: class 1:1 rate 7600kbit (on a imq device) | |\class 1:10 rate 100kbit ceil 5600kbit prio 7 (here goes p2p traffic) \class 1:12 rate 7500kbit ceil 7600kbit | |\class 1:121 rate 3100 ceil 7500kbit prio 0 |\class 1:122 rate 2200 ceil 7500kbit prio 2 \class 1:123 rate 2200 ceil 7500kbit prio

Sorry for the crosspost but it's relevant to Vorbis and Icecast I believe. I'm seeing more and more streaming stations using AACplus, with many listeners being amazed at the sound quality. Most say that a 48kb/s sounds better than a 128kb/s MP3, which would put Ogg Vorbis at around 96kb/s IMO. That means only half the bitrate is required in AACplus compared to Ogg Vorbis for the same

Sorry for the crosspost but it's relevant to Vorbis and Icecast I believe. I'm seeing more and more streaming stations using AACplus, with many listeners being amazed at the sound quality. Most say that a 48kb/s sounds better than a 128kb/s MP3, which would put Ogg Vorbis at around 96kb/s IMO. That means only half the bitrate is required in AACplus compared to Ogg Vorbis for the same