similar to: RTP Session Streaming

Hello, I am trying to create a Java GUI that will interact with an Asterisk Server. This Java GUI will essentially be a custom made SIP softphone. I will most likely use the Asterisk-Java Live API to create the connection to the Asterisk server and to open a new call. Then, I plan to use the JAIN SIP API to initialize the session and the JMF to send the audio streams via RTP when the two users

Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would

Hello, I have a small LAN network connected through an Asterisk Server (Trixbox). I was looking to create my own custom made softphones, and I have been looking into how to transmit and receive via RTP. Would anyone know how the Asterisk RTP bridging works, and if there is any documentation on it? How is the RTP stream routed through the Asterisk server? Do I just give it the endpoints and

I'm currently involved in a project where in we need to record the audio from a mic encode with vorbis, then put it in Ogg container, and then make it as RTP packet and transmit over Darwin Streaming Server. Till now we were using JMF with available codec's, since JMF handles the RTP part life was pretty much easy using it. Now we want to use Ogg Vorbis because of its adv's. So in

I'm currently involved in a project where in we need to record the audio from a mic encode with vorbis, then put it in Ogg container, and then make it as RTP packet and transmit over Darwin Streaming Server. Till now we were using JMF with available codec's, since JMF handles the RTP part life was pretty much easy using it. Now we want to use Ogg Vorbis because of its adv's. So in

Greetings: It would be nice if Icecast supported RTSP; however I would appreciate any suggestions for a small RTSP/RTP solution to encode 8kHz mono audio in GSM or ADPCM and service multiple unicast client connections. The ideal would be a black-box hardware solution with an audio input and ethernet interface similar to broadcast studio IP audio links or the network audio capabilities of certain

Hello, I have been reading up on the capabilities of the Asterisk-Java library. I believe that this library can act as an interface between a Java GUI(custom softphone) and the Asterisk server. Seems like the Live API would be easiest to use to make the connection to the Asterisk server and to set-up calls. One thing I am not sure about is how to transmit the audio streams between users'

Thomas B. Ruecker wrote: > Michael Grigoni wrote: > >>Greetings: >> >>It would be nice if Icecast supported RTSP; > > It probably never will > >>however I would >>appreciate any suggestions for a small RTSP/RTP solution to >>encode 8kHz mono audio in GSM or ADPCM and service multiple >>unicast client connections. > > why not use

Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems

Good question. I'm coming to the conclusion that using plain UDP and "home-grown" packet construction for transmitting the speex data (with timestamp/sequence counter) and implementing jitter control on the receiver end is an adequate implementation for a VoIP application. Assuming of course that I don't care about any interoperability issues with other applications etc. I was

lianghu xu wrote: > In a word, I don't what's the standard of speex payload format. > The file doc/rtp.txt is for what? Is it not for rtp payload? > I find that rtp.txt is more detail that draft02.txt > > Which rtp docment should be followed? > Anyone else has written the RTP wrapper already? Oh, I see. doc/rtp.txt was a very, very early draft. See the manual for a

Hello Opus experts! I'm new in the Opus development area - working on application that is supposed to read the opus-encoded data from the ogg opus file and send the opus media via RTP,. The RTP opus stream shall be received by some remote softphone client, decoded and stored in the PCM wav file . Could you please point me to some sample code outside of the opus and opusfile libraries

> 1) First, I think there is a bug in libspeex/lsp.c line 512. > > /* hard limit ak's to +/- 32767 */ > if (a < -32767) a =32767; // This line should be changed to if > (a < -32767) a = -32767; > if (a > 32767) a = 32767; > ak[j-1] = (short)a; Oops. Thanks for pointing that out. It's fixed in svn. > 2) About the RTP wrapper for VoIP

There's a field in the SDP description for narrowband/wideband/ultrawideband. Jean-Marc lianghu xu wrote: > if the new draft in the manual is used. I don't find how to tell the > decoder which mode(NB/WB/UWB) is used > in the encoder. The RTP header don't contain the mode field and I don't > find the mode information in the > coded frame either. > >

hi .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

hi, I use the oh323 driver to answer H323 calls. The connection is set up normally. In my extensions.conf file I use: exten => s,1,Answer exten => s,2,Playback(demo-instruct) exten => s,3,Hangup So that when a call is answered i get: *CLI> -- Executing Answer("H323/ip$10.0.3.23:32782/6502", "") in new stack -- Executing

In a word, I don't what's the standard of speex payload format. The file doc/rtp.txt is for what? Is it not for rtp payload? I find that rtp.txt is more detail that draft02.txt Which rtp docment should be followed? Anyone else has written the RTP wrapper already? Lianghu On 11/22/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > > > 1) First, I think there is

Hi Jean-Marc and All, Two things need your confirmation/suggestion. 1) First, I think there is a bug in libspeex/lsp.c line 512. /* hard limit ak's to +/- 32767 */ if (a < -32767) a =32767; // This line should be changed to if (a < -32767) a = -32767; if (a > 32767) a = 32767; ak[j-1] = (short)a; 2) About the RTP wrapper for VoIP I'd like to use the

On 03/11/03 00:25, Mark Spencer wrote: > As a side note, I strongly would like to see someone implement a > client using libiax2 which implements IAX2 instead of the (now > obsolescent) IAX version 1. I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java

I am a brand new user of Speex, so forgive my naivety. I am receiving an audio stream (PCM) which I need to encode in Speex, send it across a wire via RTP, and then decode and play the audio on the other side. I assume people have done this before, and I was wondering if anyone had any sample code of how they accomplished this. Or maybe just some good pointers or advice. Thanks Shane