RTPJobb

...streaming client server application. Application Structure: Web based Encoding Client --> Streaming Server --> Flash player or windows media player The application needs to have components as follows: - Encoding client : web based streaming client to capture audio/video, encode it and sends it to the server in RTP, UDP and h.264 format (like WME or web based encoders of ). - Media Streaming Server :streaming server to get RTP/UDP stream and Make the h.264 stream available for live view that can be viewed by 100 flash players hosted on a webpage on the server or play in windows media player. Requirements: - Live video Delay should be <1 second. - Video should not freeze. - Audio/video must be in sync. - CPU usage on encoder/viewer sho...

MIDI encode/decode RTP-MIDI/SIP client Web/SIP server Video sharing (ytube like), Craigslist like classifieds postings/viewing, Live Video/ Conference chat support Live conference Audio support. If you have readily available solutions to any of the features please contact US ASAP

...a client server application that will allow IP based RTP/UDP video/audio conferencing between 25 people. You should be using c# (preferred, otherwise c++ also ok as long as APIs can be called from c#). Idea like as follow: RTSP Client (Sending part): Sending Filter Graph at A is composed of: 1-Audio Capture filter to capture audio from the microphone. 2-Video Capture filter to capture video from webcam. 3-PCM wrapper to convert the data to PCM. 4-speex transform filter to encode the PCM data into configured audio codec. 5-Theora filter to encoder the video data into configured video codec. 6-RTP Send Filters RTSP Server: 1-RTP Receive filters. 2-Mux Filter. 3-RTP Send Filter. RTSP Client (Receive part): 1-RTP Receive filters...

...enhance existing streaming client server application for RTP,UDP and H.264 with any/all of following references: ** Following are few of the the application whose source can be referred for the purpose of this project: 1) <> 2) <> 3) <> 4) <> 5) <> 6) <> The application needs to have components changed as follows: - Encoding client Enhancement: streaming client to capture audio/video, encode it and sends it to the server in RTP, UDP and h.264 format (like WME or encoders of

...Directshow expert to build streaming client server application in form of ActiveX and XPI (IE & Firefox) for RTP,UDP and H.264 with any/all of following references: ** Following are few of the the application whose source can be referred for the purpose of this project: 1) The application needs to have components as follows: - Encoding client: --> VLC activex and xpi acting as encoding client should capture audio/video (webcam/microphone), encode it and sends it to the server in RTP, UDP and h.264 format (like WME or encoders of ). - Media Streaming Server: --> VLC activex and xpi streaming server should get RTP/UDP stream and Make the h.264 stream available for live view that can be viewed by 100s of flash and silverlig...

c#/c++Directshow expert to enhance existing streaming client server application for RTP,UDP and H.264 with any/all of following references: Following are few of the the application whose source can be referred for the purpose of this project: 1) 2) 3) 4) 5) 6) The current application has following components for audio/video streaming: Encoding Client --> Streaming Server --> Windows Media Player The application needs to have components changed as follows: - Encoding client Enhancement: streaming

**c#/c++Directshow expert to build RTP/UDP Video streaming client server application with any/all of following references: ** 1) <> 2) <> 3) <> 4)? <> 5) <> 6) <> The application needs to have following components: - Encoding client: RTP/UDP streaming client? captures audio/video and encodes it and sends it to the server in h.264 format (like WME or encoders of ). - Media Streaming Server: RTP/UDP streaming server that Makes the h.264 stream available for live view that

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**c#/c++Directshow expert to build RTP/UDP Video streaming client server application with any/all of following references: ** 1) <> 2) <> 3) <> 4)? <> 5) <> 6) <> The application needs to have following components: - Encoding client: RTP/UDP streaming client? captures audio/video and encodes it and sends it to the server in h.264 format (like WME or encoders of ). - Media Streaming Server: RTP/UDP streaming server that Makes the h.264 stream available for live view that

Looking for a turn key project to deliver the following solution: one or more from these list of mobile platforms: 1. Symbian 3rd edition / Symbian UIQ 2. Iphone. 3. Windows mobile. 4. Android. 5. BlackBerry. For each platform the client should have the following features and capabilities: 1. SIP 2.0 fully compliance for signaling. 2. RTP with G.711 A/U-LAW, G.729AB. 3. WLAN & 3G infrastructure usage. 4. Dialer can run behind NAT or on private IP. Dialer supports NAT and firewall traversal using STUN. So client behind the firewall can make VoIP call using Dialer . 5. Auto start / Auto connect. (preferred) running in phone background 6. Secured remote provisioning: install ,update, delete. 7. Dialer Supports Loud Speaker 8. SMS & IM 'Yahoo,MSN,Skype...

...other SIP soft phone or any landline or mobile number via a VOIP service provider of our choice. The phone is implemented as a java applet/application and it is completely independent platform running on webpages, windows desktop, MAC, Linux, Solaris and mobile devices. It can be used as a normal softphone running on our website or as Skype-like buttons (Click to Call). Features: SIP and RTP stack (compatibile with standard VOIP servers like Cisco or Asterix, FreeSWITCH, etc) Standard java applet (no installation required. runs directly from browsers) Standard G711 codec’s (PCMU and PCMA) and speex narrowband DTMF (INFO method in signaling) Basic call features IM (chat) capability based on SIP SIMPLE protocol Other VOIP related features will be coming soon...

this project is for developing an sip engine for nokia s60 platform (Fp1). the dialer should be implemented with the nokia sip stack. ICE, STUN support is needed. also multiple accounts registration/multiple sip calls should be handled. no need for rtp stack. a test sw that uses the engine to make calls and pass rtp packet between the peer should be included. attach UML with the desired interface to the engine. the uml is just the basic idea of the interface.

...information, portfolio with mobile / voip projects. Looking for a turn key project to deliver the following solution: one or more from these list of mobile platforms: 1. Symbian 3rd edition / Symbian UIQ 2. Iphone. 3. Windows mobile. 4. Android. 5. BlackBerry. For each platform the client should have the following features and capabilities: 1. SIP 2.0 fully compliance for signaling. 2. RTP with G.711 A/U-LAW, G.729AB. 3. WLAN & 3G infrastructure usage. 4. Connection management with network priority list - register to the service by preferred network priority list e.g - if not under WLAN coverage connect to 3G, when arriving to favorite WLAN converge disconnect 3G and connect to WLAN. 5. Auto start / Auto connect. 6. Secured remote provisioning: instal...

RTP Stream Server I need an RTP stream server compiled for Linux with the following features: 1) The server will need to have user authentication, It will need to read a Mysql database with a list of username and passwords and only allow access to a stream to those users; this could be done via unique URL’s containing the users username and password: EG: RTP://(ipaddress)/(portnumber)/(User/Password)/Filename 2) The server must be able transmit multiple streams (from the same, or different files) concurrently. 3) The server must be able to stream files that are stored in a local directory. I have had a look at Live555 media server and it comes close to meeting our requirements. I think that this project could be achieved by compiling

this project is for developing an sip engine for nokia s60 platform (Fp1). the dialer should be implemented with the nokia sip stack. ICE, STUN support is needed. also multiple accounts registration/multiple sip calls should be handled. no need for rtp stack. a test sw that uses the engine to make calls and pass rtp packet between the peer should be included. attach UML with the desired interface to the engine. the uml is just the basic idea of the interface.

RTP server using JMF for G711 I need a skilled Java developer to compile a custom install for JMF It will be used to stream G711 audio VIA RTP and will need the following features 1) Will need to be able to receive/read multiple static files hosted on the same server in G711 format then output these files Via RTP . ***Must be able to support multiple inputs and outputs and pass output streams to different port numbers**** 2) Will need support user authentication: the server will need to periodically read a database of users and create authenticated streams for those users; so each user’s link to the rtp stream will be different and contain their username and password (example HTTP://ipaddress/port/user/ or some thing like that – cant reme...

I am looking for consultence about using pjsip. what i need is to use pjsua for the signaling, this part is already covered. the project is about getting hooks on the basic rtp channel so that i can use the channel without using the hole pjmedia lib.

I need a Directshow? RTP Sender and an RTP Receiver filter for for both Audio and Video. The filters will be very basic to the extent that they will only ever? need to? parse h264 encoded bytes? for the video and AAC encoded bytes? for the audio. The encoding and decoding will be handled outside of these filters. The handling and checking? for multiple file types will not be necessary. A typical graph will be as follows: SourceFile->Splitter/Demux->h264Enc/AACEnc->RTPSender. RTPReceiver->Splitter/Demux->Video/AudioRenderer. The PropPage will only need to allow for Multicast IPAddress and Port. NIC will be any. I will also need the source code and a reference of the calls that i will need to make from C# when adding the filter t...

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...is around 10 seconds, the underlying codec is wmv, By replacing wmv with h.264, the video delay between encoder-->server-->viewer should be brought to less then 2 seconds. 2 - The server right now works when the encoding client is in the same network...when server is running on live IP and client is behind the NAT, it doesn't work. Please fix this by converting TCP based application to UDP/RTP (this should fix issue 1 as well). It should be 2-4 days task for somebody who knows what he/she is doing. To be given: Existing working encoder & server. Deliverable: Updated Encoder, Server and Updated Player (either custom directshow filter for WMP or VLC) Let me know if any queries. ## Deliverables 1) Complete and fully-functional working progr...

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we are looking experts in the area of VOIP. We have spent 2 years developing a PC to PC softphone and now looking add video conferencing facility and would also like to fine tune our media handler for better audio quality. i am looking for people having extensive knowledge in SIP, RTP and Handling Audio/Video Codecs + extensive knowledge about UDP ports. Thanks

I am looking for symbian S60 developers. The work will be done on each module/engine basis. there are several engines to design and code: 1. SIP - based on pjsip or nokia internal sip implementation. 2. Voice engine - based on VAS (voip audio services) and jitter buffer implementation. 3. RTP Engine. 4. Connection meneger. 5. Encryption engine - engine to do cryptographic calculation in the background. 6. Speex integration. 7. GUI. 8. Db, Contact meneger, etc. the developer should be expirience in one of the above and in symbian S60 in general. the app should pass symbian signing creteria. the work is done by fixed price per engine AFTER agreeing on the desired interface and testing procedure.

...app must support multiple codecs such as G711, G723, G729, ALAW, ULAW, iLBC 8. the app needs to work behind firewalls or blocked ports 9. the app needs to communicate with standard VOIP Providers supporting the SIP Protocol the application needs to have good voice quality and may or may not include all or some of the industry standards and protocols below: - UMA - IETF - IMS - SIP - RTP, RTCP - STUN - SDP - codecs support G711, G729, G726, iLBC, ALAW, ULAW i will require the source code, files & all Intellectual Property and copyright for this job. To be considered for this project, you need to answer these questions and post thru the GAF system: 1. What IDE do you intend to use 2. What API, library or SDK is to be used 3. Confirm that your bid price...

...app must support multiple codecs such as G711, G723, G729, ALAW, ULAW, iLBC 8. the app needs to work behind firewalls or blocked ports 9. the app needs to communicate with standard VOIP Providers supporting the SIP Protocol the application needs to have good voice quality and may or may not include all or some of the industry standards and protocols below: - UMA - IETF - IMS - SIP - RTP, RTCP - STUN - SDP - codecs support G711, G729, G726, iLBC, ALAW, ULAW i will require the source code, files & all Intellectual Property and copyright for this job. To be considered for this project, you need to answer these questions and post thru the GAF system: 1. What IDE do you intend to use 2. What API, library or SDK is to be used 3. Confirm that your bid price...

... by a security ip camera, so it can be seen on any security program. Currently, the webcam stream is received, some text is written on the stream and it is transmitted over the network in JPEG (or RGB not sure) using RTP Protocol. As far as I understand what it is needed is to convert the stream, on the fly, from its current format to mjpeg, the format used by most ip cameras. However, it might not be the right solution and I would like to listen any other possible solution.? The solution could be using java and converting before sending, or capturing the rtp stream in the machine itself and convert it before sending it againg, using any program written in any language.? The software must run in linux, debian distribution.? ## Deliverables The softwa...

c/.NET/vc++ SIP programmer required to develop SIP client for voice and video calling with Windows Mobile 6.1 and wifi/3g/edge. We have complete SIP/RTP desktop client stack implementation in c# based on codeproject sample which would need to be enhanced. Implement audio/video codecs to support audio and video calling ...the stack is a complete SIP and RTP stack client that supports calling. But, does not support voice and video yet. So, support for following codecs need to be added: Audio: gsm Video: h.264 Requirements: - Address book - audio/ video call - login/ register - Audio/ video must be synchronized and real-time without delay. Quality: benchmark for 244MHz & 42MB RAM Mobile - 320x240 video size, 4-5 fps. Deliverables: Work...

A 3 way teleconferencing program to be developed using either SDS or WebLogic along with JMF ( SIP and RTP need to be written as well) for client program you can use xlite oy thing similar: the program should work the following way: - on a small lab, on 3 computers, 3 parties can join a live audio conference with invite and finish capabilities. -can be kept for as simple as possible -shouldnt be long, pehaps 10-20 hours of work -SDS or Weblogic configurations maybe needed.

SIP programmer required to develop SIP client for voice and video calling for android. We have complete SIP/RTP desktop client stack implementation in c# based on codeproject sample which could be used and enhanced. Implement audio/video codecs to support audio and video calling ...the stack is a complete SIP and RTP stack client that supports calling. But, does not support voice and video yet. So, following codecs need to be supported: Audio: gsm Video: h.264 Requirements: - Address book - audio/ video call - login/ register - Audio/ video must be synchronized and real-time without delay. Minimum Quality benchmark for 244MHz Mobile - 320x240 video size, 64kbps bitrate, 7-10 fps. Deliverables: Working solution with source code. 2 people be able to do video c...

SIP programmer required to develop SIP client for voice and video calling for symbian. We have complete SIP/RTP desktop client stack implementation in c# based on codeproject sample which could be used and enhanced. Implement audio/video codecs to support audio and video calling ...the stack is a complete SIP and RTP stack client that supports calling. But, does not support voice and video yet. So, following codecs need to be supported: Audio: gsm Video: h.264 Requirements: - Address book - audio/ video call - login/ register - Audio/ video must be synchronized and real-time without delay. Minimum Quality benchmark for 244MHz Mobile - 320x240 video size, 64kbps bitrate, 7-10 fps. Deliverables: Working solution with source code. 2 peop...

Hello everyone this is small java project to make video streaming, please check the attachment for more details.

SIP programmer required to develop SIP client for voice and video calling for symbian, android, mobilin, iphone and windows mobile. We have complete SIP/RTP desktop client stack implementation in c# based on codeproject sample which would need to be enhanced. Implement audio/video codecs to support audio and video calling ...the stack is a complete SIP and RTP stack client that supports calling. But, does not support voice and video yet. So, following codecs need to be supported: Audio: gsm Video: h.264 Requirements: - Address book - audio/ video call - login/ register - Audio/ video must be synchronized and real-time without delay. Deliverables: Working solution with source code. 2 people be able to do video call with each other having...

...stream from flash has to be converted to the RTP VoIP stream. - In addition to that the corresponding SIP signalling to a VoIP application server has to be established. Explanation: The VoIP protocol consists of a SIP signalling text based protocol which used port 5060, and a corresponding audio stream (RTP stream) that uses higher ports between 30000 - 60000 - The process also has to work the other way around. If an audio VoIP call is received on the Asterisk VoIP server it is send to the red5 server that has to convert the call to a flash media call. - The implementation should be based on the following open source components: Red5 (Java Flash media server), Java SIP Protocol stack (e.g. Jain), Open Source RTP stacks for Java (just search for RTP/j...

An existing project receives rtp streams via a network from several ip cameras and displays them inside a security application. The project uses a DirectShow filter chain based on a modified DSPack push source filter followed by elecard/mainconcept's push demultiplexer to detect the video stream format (mpeg-4, h.264 etc.). After detecting the format the application connects the decoder and renderer. The filter part is programmed in Delphi. Now we want to improve the filter graph by eliminating the push demultiplexer and connecting the source filter directly to the decoders (eventually ffdshow).

Streaming video over RTP using already existing SDK. I want help to build some exe files from my .grf (graphedit). Graph consists of SDI input source OR ms dv capture source, encoded in real time to h.264/AAC, muxed RTP and sent as unicast. I just don´t have the programing knowledge/time to do it myself.

The project is to enhance the windows voip SIP pbx stack for client/server (similar to asterisk) as follows- Step1: Implement audio codecs to support audio calling ...the stack is a complete SIP and RTP stack server/ client that supports calling. But, does not support voice and video yet. So, following codecs need to be supported: Audio: g729, g711, gsm, ilbc, g723 - Allow various clients like xlite/ eyebeam and the built in SIP client to register with the sip server - Allow 1-1 voice calls to happen between these clients. Call should be of high quality. Deliverables: working source with updates as required. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-...

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I have a javascript program that performs simple calculations and has a function to save the results to a .csv file. The save function is no longer working properly. The csv file does not contain any of the input data that it should. I require: The script to be fixed so that the csv file contains the correct data. A ...javascript program that performs simple calculations and has a function to save the results to a .csv file. The save function is no longer working properly. The csv file does not contain any of the input data that it should. I require: The script to be fixed so that the csv file contains the correct data. A log of changes made to the file(s). You can view the script at I will attach also the files to this project.

...detailed explanation document is available to the selected provider. All reports are to be build using XSD as datasource for the rpt file – I already have a very simple tool to generate the XSD file from SQL queries. As a summary, to build each report you will have to: 1) Build the query(ies) you need to use in the report. 2) Create a XSD file to use as the Crystal Report report file (rtp) datasource. 3) Create a Crystal Report file (rpt). 4) Create a XML file. 5) Save files to the server. 6) Test it! I am attaching an excel sheet with a specification example I send for each report. Important: If you are not an experienced professional who has proved know-how in this type of technology, if you are not used to build professional looked reports, if you do ...

The project is to enhance the windows voip SIP pbx stack for client/server - (similar to asterisk) Requirements set 1 listed here: Step1: Implement audio/video codecs to support audio and video calling...the stack is a complete SIP and RTP stack server/ client that supports calling. But, does not support voice and video yet. So, following codecs need to be supported: Audio: g729, g711, gsm, ilbc, g723 Video: h.263, h.264 Step 2: Achieve Basic features listed here: Step 3: Achieve Advance features listed on the same page. Project involves working with more or less 9-10 c# files only. 10 days task for somebody who knows what he/she is doing. This project is urgent and one of the many in pipeline

Take an MP3 file convert to RTP packet(s) and send over network. ## Deliverables Basically I need to parse an mp3 file and convert it to RTP packets. This should be done in a fashion so that a server can receive the RTP stream and reconstruct the MP3 file. This may mean stripping the MP3 Headers from the file???? I need a class or classes that will represent RTP packet(s) and a test harness that will send the MP3 to a defined IP and Port. Built in VB 2005 or Preferable in VB 2008. Ducumentaion in the form of extensive comments in the code.

Thanks you for looking at my project... I want to be able to distribute audio programming over the Internet with as low a latency as I can get ie. RTP protocol. What I am looking for is a simple audio server application that can relay audio from a source and distribute it to multiple client units using RTP. I have purchased two pieces of hardware from Barix, the Instreamer and Exstreamer. The Instreamer is the encoder and transmitter. This units takes the audio and encodes it and streams using RTP. The Exstreamer is the receiver that receives the data and decodes it into audio. The Instreamer will encode and stream it over the Internet to this piece of software, installed on my Windows 2003 dedicated server, then the Exstreamers (clients) will receive it from this so...

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? Video conferencing application using webcam, implementing H264 algorithm for frame compression and application will provide a way to connect peer-peer system via internet/intranet for video conferencing. The scope of the application is limited to video/audio conferencing only. Since this application uses RTP implementation, any number of users can join/leave the video conference. This application will act as a RTP server/client. The application should be developed in C# time protocol will be used for fast transmission of compressed video encoder/decoder should use the H.263 or H.264. Also FFMPEG is packets will be sent using can start/stop video conference at frame size is configurable by user. Allowed sizes are,

We are looking for company who can develop audio capture on dsp - encoding to AAC-v2 over rtp using VC++ or c++. Details of development will be given to qualified bidders who have g.711/rtp skills. We would like this work completed very fast.

We are looking for company who can develop audio capture on dsp - encoding to AAC-v2 over rtp using VC++ or c++. Details of development will be given to qualified bidders who have g.711/rtp skills. We would like this work completed very fast.

Our Videolan server will act as a DVB-S to IP gateway and redirect the satellite channels Windows Media Encoder. But Windows Media Encoder (WME) doesn't support to receive the stream from network. Can you build a component for WME so it can support to receive the stream by RTP or HTTP source from Videolan? The network stream should be showed as Video and audio device on Windows Media Encoder. Please send me offer and delivery time, tell me if you have worked in such projects before. The delivery time should not be more than 1-2 weeks. I want to begin work with the project TODAY!

I am needing a program created to run on a mobile phone (phone types to be discussed) which can stream live video and sound footage over a 3g connection from the phones camera using the RTSP/RTP h.264 protocol I have stated Symbian C++/J2ME in the title, but if you feel you can do it in any other language then that is fine with me.

Hello I have installed vlc on a debian etch server (32bit). Now I want to use it for video-streaming. I have about 4-5 videos that should be played in ...file. Furthermore I want to change the videos from time to time without huge effort. What I expect is a fully running and configured VLC. The following commandline should work and put the stream into the .sdp-file: (found in a forum) vlc --sout '#duplicate{dst="transcode{width=640,height=480,venc=x264,vcodec=x264,vb=300,acodec=mp4a,ab=128,channels=2,samplerate=44100}:duplicate{dst=rtp{dst=,port-video=10000,port-audio=10002,sdp=file:///var/streaming/content/}}"}' vlc:quit For realizing this job, you get root access. I also expect a short documentation listed all steps you did. Best regard...

Deliverables:$0? ? Oreka (<>) built for CentOS 5.$0$0? ? Source code.$0$0? ? Step-by-step guide of compilation procedure.$0$0? ? Sample audio files proving the system works.$0$0$0$0$0Requisites:$0$0? ? Just like the original, it must be able to record RTP traffic into wav audio from a second network card in the computer.$0$0? ? Must be able to capture pertaining SIP information to associate with the generated wave file (a dump of the SIP headers would suffice) in a same named file (ie & ).$0$0? ? Must not use any non-open source components as the resulting sofware will be released as open source.$0$0$0$0$0Answers to possible question:$0$0? ? Yes, it must be on a CentOS 5 platform.$0$0? ? I am flexible on the audio format. Wav, au, mp3. Must