Codec sipdroidPekerjaan
hello i need a debian linux(vlc and ffmpeg installed only) which must pxe boot x264 / AVC Codec (libav) must be installed.. new codecs must be installed. simple web server with php support. and a cron to run after reboot once.
...share computer desktop with anyone. Presenter only need one executable program Viewers do not need to download or install anything and need supports all platforms, must use a flash application. Presenter software need be a toolbar on desktop top similar to Presenter software need use a color quantization process Presenter software need use a jpg compression/decompression codec like libjpeg-turbo or better Presenter software should work across all firewalls Presenter software must create a unique join url for Viewers. Presenter screen packets must be in high quality colors Presenter need publish screen/data packets to a Wowza Streaming Server Viewer flash application should run in a webpage and use request string parameters. Viewer flash applica...
Libavg:- Libavg is a open source cross platform multimedia framework which uses python as scripting language and is written in high-speed C++. The C++ core is written with performance in mind, all graphics processing is hardware-accelerated through OpenGL, and GPU Shaders are used to accelerate critical parts like video decoding, compositing and visual effects. Currently, accelerated video decoding is supported using NVidias VDPAU interface. In the case of VDPAU, libavcodec will do the decoding on suitable hardware but expects to output the video directly to the graphics card. It would be really great to support VA-API (Intel's counterpart for VDPAU) on compatible hardware from libavg. libavcodec already supports both but there would be some refactoring work to be done inside liba...
Project Subject: Android Live picture filter & broad cast Overview : Application need to capture stream video , then grab each frame of video stream , apply the filter / panoramic correction / banana codec on each frame , finally broadcast the stream video on TCP/IP URL or save it in local storage ( internal memory / SD card ) . ______________________________________________ Diagram: Please find the attached diagram image . ______________________________________________ Description: There are three step in this project according to the diagram . below we explain & show sample project as example to understand it better . STEP1: Using smart device camera to Live Camera preview or surfaceview SurfaceView Android Class could be used for this purpose or I...
...multiple DLE - Auto Publish post to multiple IPB Forum - Auto Publish post to multiple PHPBB Forum - Auto Publish post DRAFT OR Publish Imdb Graber Auto Get Cover from NFO IMDB URL Auto Get Full Info From NFO IMDB URL Movie - Download Torrent (on vps) - Unrar If rar (on vps) - Get Cover From NFO IMDB URL - Get Full Info From NFO IMDB URL - Creat ScreenShot From Video - Creat Full Video Detail Size Codec Runtime Your Sitename NFO File - Rerar NFO + Video File - Upload To Filehost - Upload Screenshot to Imagehost - Get Download Link - Creat Complete Post - Sent Post ON wordpress Supported File Hosts : In special , , , , , , , , , ,
We require a SIP Dialer for iphone & android phone with chat features based on XMPP chats, life gtalk, yahoo, aim, hotmail, facebook & others if possible login page asking for username & password. SIP Server ip address, codec and some other configuration is hidden. Embed with phone book contacts detection, phone's SIM number detection. White label dialer required with our company name & logo. Codec - G729 required.
We require a SIP Dialer for iphone & android phone with chat features based on XMPP chats, life gtalk, yahoo, aim, hotmail, facebook & others if possible login page asking for username & password. SIP Server ip address, codec and some other configuration is hidden. Embed with phone book contacts detection, phone's SIM number detection. White label dialer required with our company name & logo. Codec - G729 required.
...Operating System OS. C. Apart from all normal feature of the cell phone it must have the APPLICATION with built in encryption and tunnel FEATURE which will register the cell through a LOGIN and PASSWORD to a compatible Soft-switch through internet by SIM1 D. It must have a MY-SIP DIALER or such dialer that will use SIM2 to connect the desired number coming from the SoftSwitch 5. A very good voice codec must be develop which can delivery high voice quality from SIM2 ----> SIM1 ----> Softswitch -----> Origination side with less bandwidth consumption and/or capacity. Necessary test must be done to measure the bandwidth consumption is minimum which is very very important here. 5. A soft-switch preferably built in ASTERISK with mysql which will be able to do must have ...
Create something familiar with : Features that needed are : Video Conference key features + many to many 2 way video chat + 3 default video panels + unlimited extra video panels + user list with online participants + room list + file sharing + private messaging + P2P group streaming support + codec setup (H264, Speex, NellyMoser) + acoustic echo cancellation + session timer and control with scripts + 100% web based for clients Developer will hand over all the files and the source code.
this project is similar to www dot getacoder dot com/projects/symbian%20blackberry%20sip%20dialer_140570 dot html. I am looking for a symbian SIP dialer which supports g729 passthrough codec with sip server ip hardcoded in it . Ready made work welcome as long as you can provide source code with compiling instructions . Please bid only if you have experience . netcallsbabu Skills required: Nokia, Symbian
this project is similar to www dot getacoder dot com/projects/symbian%20blackberry%20sip%20dialer_140570 dot html. I am looking for a symbian SIP dialer which supports g729 passthrough codec with sip server ip hardcoded in it . Ready made work welcome as long as you can provide source code with compiling instructions . Please bid only if you have experience . netcallsbabu Skills required: Nokia, Symbian
this project is similar to www dot getacoder dot com/projects/symbian%20blackberry%20sip%20dialer_140570 dot html. I am looking for a symbian SIP dialer which supports g729 passthrough codec with sip server ip hardcoded in it . Ready made work welcome as long as you can provide source code with compiling instructions . Please bid only if you have experience . netcallsbabu
...module for test. h) You need use my DIGIUM TDM800P (8fxo). i) all asterisk sounds in Brazilian Portuguese. You need find this sound in internet. j) I want use elastix. k) all config need work in Brazil. l) Text can to stay in english. m) i need change URA config day/night in manual mode. n) for each voice mail message the system need send one email message notification. o) i want use g729 codec p) I will not keep this job personally. I will appoint one developer to accompany this work. You will communicate with my developer using our portal at developer.registrocom.com.br. Our developer team work from 00:00PM to 06:PM BRST (Monday to Friday) II - INFORMATIONS At the end of project all password need change. Will be necessary to inform: a) How to change extension...
My client needs to use an android cellphone as an asterisk channel/Gateway. His aim is to make phone calls using his Asterisk PBX through a cellphone running android. You are free to choose how to im...aim is to make phone calls using his Asterisk PBX through a cellphone running android. You are free to choose how to implement this feature. All that is required is that in the end the Android phone will be acting as a SIP or IAX to GSM channel/trunk/FXO and other SIP devices can connect and make phone calls using this. You choose the best way to do it: Add the feature to SIPDroid: Add the feature using the BatPhone Asterisk Build: Or do it your own way.
Project Description: We need a Voip client for android based on the following. 1. To view the functionality of the program you may download the program from play store CSipSimple (by Regis Montoya). 2. You can find information and SDK NDK tools at the following. 1. To view the functionality of the program you may download the program from play store CSipSimple (by Regis Montoya). 2. You can find information and SDK NDK tools at 3. We need an application that will ask the customer Account Name (Will be the same as User) and password. This application will be locked to our Server The codec will be locked to GSM (See settings -> Media -> Codecs -> Choose GSM ) 4. Add a Logo of our company.
We need a Voip client for android based on the following. 1. To view the functionality of the program you may download the program from play store CSipSimple (by Regis Montoya). 2. You can find information and SDK NDK tools at 3. We n...following. 1. To view the functionality of the program you may download the program from play store CSipSimple (by Regis Montoya). 2. You can find information and SDK NDK tools at 3. We need an application that will ask the customer Account Name (Will be the same as User) and password. This application will be locked to our Server The codec will be locked to GSM (See settings -> Media -> Codecs -> Choose GSM ) 4. Add a Logo of our company.
We require a SIP Dialer for iphone & android phone with login page asking for username & password. SIP Server ip address, codec and some other configuration is hidden. Embed with phone book contacts detection, phone's SIM number detection. Current white label dialer accepted.
...share computer desktop with anyone. Presenter only need one executable program Viewers do not need to download or install anything and need supports all platforms, must use a flash application. Presenter software need be a toolbar on desktop top similar to Presenter software need use a color quantization process Presenter software need use a jpg compression/decompression codec like libjpeg-turbo or better Presenter software need use a video driver mirroring like Presenter software should work across all firewalls Presenter software must create a unique join url for Viewers. Presenter screen packets must be in true colors Presenter need publish screen/data packets to a Wowza Streaming Server Viewer flash application
...Following features must be available in the dialer: Making & Receiving VoIP Calls Over 3G, WiFi, GPRS. -Video calling -Voice calling -Messaging -Filesharing The dialer should also provide Vpn/tunneling support for working in SIP/Voip blocked countries. Mobile Dialer should support SIP/H323 for signaling and should work with any standard SIP server. Mobile Dialer supports G729, G723.1 and G.711 codec for sending audio stream. Mobile Dialer can run behind NAT or on private IP with VPN Tunneling embedded. Integrate with phonebook contacts Dial, Hang up, Redial Call Hold Call Mute Call Forwarding Call Waiting Speaker Phone Keypad Display Call History: All, Recent, Dialed, Missed Account Setup Work with all standard SIP servers Supports all standard SIP suppor...
...SIP/IAX2 Client: SIP/IAX2 client list, Add new SIP/IAX2 client, Edit SIP/IAX2 client, Delete selected SIP/IAX2 client,? Inbound Client: Inbound client list, Add new inbound client, Add IP for inbound authentication, Edit inbound client, Delete selected client,?? Outbound Trunk: Outbound trunk list, Add new gateway trunk IAX2/SIP/IAX2 with Trunk name – IP Address – SIP/IAX2 port and Codec selections, Edit outbound trunk, Delete selected trunk,??Dial Plan: Dial plan list, Add new dial plan with Dialing prefix – Dialing rules – Priority – Outbound Gateway selection , Edit dial plan, Delete selected dial plan, ???CDR: Call record report, Search field (Date from and to - Time from and to – Client - Gateway) Display record (Call date and Time &nd...
I need a...and video streams from a single video file (e.g. ) into separate audio file (e.g ) and video file (). Requirement: 1. Written with Visual C++ for Windows 64 bit machine. 2. Utilize ffmpeg () library. [Note: you need to use the windows x64 version of ffmpeg. ) 3. No particular requirements on video/video codec or file format. 4. No GUI is needed. Deliverables: Complete Visual Studio 2010 solution with all source codes, including ffmpeg library (/include,/lib,/bin). This is a feature already supported in FFMPEG. So it should be very straightforward to implement. Useful hint: It probably only needs a total of 100 lines.
...property for these two GUI-less win32 command line programs. In Visual C# or Visual C++. And in worst case VisualBasic 1. - records a videostream with autodetected codec to a file. : URL File :c: rtsp://AxisIP/axis-media/ c: And a mechanism for ending the recording. For a file "" is present, after is started, in the same folder as it will stop the recording and exit within 3 seconds. 2. - takes a snapshot from the videostream with autodetected codec to a file. Not via http://AxisIP/jpg/1/image.jpg. It has to be from the videostream! : URL File :c: rtsp://AxisIP/axis-media/ c:video
SIP Mobile Dialer (iPhone, Android and PC) with VPN Tunnel: It Should work from any block or unblock country as UAE, OMAN, KSA etc. Customized design with our company logo. Use SIP and H.323 for signaling. Compatible with all Soft Switches, supporting SIP. User friendly interface. Flexible integra...our company logo. Use SIP and H.323 for signaling. Compatible with all Soft Switches, supporting SIP. User friendly interface. Flexible integration with your Phone Book contacts. Support Incoming & Outgoing call. QoS for VoIP Calls. Display Balance. No extra Balance server is required. Dialed/Received Call History. Call timer. Refill Account. Support G729, AMR, ILBC, GSM, Speex, G711 codec Compatible with EDGE, GPRS, Wi-Fi & VPN Access points NAT Traversal or ...
...server Main goal minimize Bandwidth in client side with quality voice . Required bandwidth compression (upto 60-80% in reality then theory) from Server X to Server Y. A usual SIP in G729 call takes 27-32kbps per port Explanation of scenario: 1. Server X (asterisk server, with static IP) receiving VoIP calls from different Carrier/Originator, with h323/sip protocol, using G711,G729 and/or G723r6.3 codec and sending calls to Server Y. 2. Server Y {Windows based Asterisk server with PRIVATE NETWORK IP, receiving calls from server X and sending to gateways (quintum, goip etc brand) or E1 cards. (Please take a look at this page for better clarity ) 3. Number of Server Y can be unlimited. 4. Number of Gateways/E1 cards per server
Hallo i wanna own player video thats work with wordpress ! own player video mean , it can play any video codec. like avi,mp4,flv.....ect a little icon thats show each video . volum and full skren and those normal think should be on . simple and eazy player thats visitors can play it . share , facebook, imped ,,,ect work with iphone and ipad (if it will cost,i wont it) waiting for ur offers guys . i looking for good price and kvalitet ! Thanks
Definition 1) There’s a x264 plementation of h.264 codec that we’d like to use in our own product. 2) There’re several issues with current x264 that prevents us from using it out of the box It’s distributed as source files (mainly for Linux) and we need to port it to Windows and to be represented as Visual Studio 2008 project It doesn’t have Direct Show filter available. What needs to be done 1) Port source code of x.264 to Visual Studio 2008, make it compiled, built out there 2) Develop DirectShow filter to encode video stream and to be able to Configure encoding parameters This can be done on top of existing implementation Success criteria 1) DirectShow filter to enable x...
...User friendly interface 2) Mobile Dialer should support SIP/H323 for signalling. 3) Mobile Dialer supports G729, G723.1 and G.711 codec for sending audio stream 4) Mobile Dialer can run behind NAT or on private IP with VPN Tunneling embedded 5) This software will run on the phone having Symbian or Windows Mobile 5 and 6 having internet via GPRS/EDGE/Wi-Fi/3G and allother available OS and Phone models. 6) Mobile Dialer should be fully customizable. 7) Able to use Mobile Dialer with any kind of Softswitch,Voip Billing Platform which support SIP/H323. The dialer must be runing from the application layer. The most important part is to add G729 and G723.1 codec in it. Second important part of this project is to add tunneling feature so call can placed even where VOI...
Hello, I need this software on my Debian-6 dedicate Server, I have the SKADATE Dating Software Script and need this tools: - Project Resources: FFmpeg () Mplayer + Mencoder () Flv2tool () Libogg + Libvorbis () LAME MP3 Encoder ()
I want to add codec G729 to increase quality call on my system and make a modification on my server about sell rate to remove in reseller panel.
...(chromakey/green screen) with a given background video (same length, framerate, format, codec) to a result video (incl. sound) in save in a target folder (ini); the sounds of both videos should be mixed at same level each. the videos will not be longer than about 30 sec, if any additional parameter needed in ini file, lets discuss. As i am not available every day an anser can take 1-2 days. As first milestone i see a working version with a container/codec combination you like. I pay for this lets say 30 %. A final release contains all named container/codecs, is well tested and deliverd with sources. Platform is Windows XP / SP4 english. ini file: - source folder - target folder - may be container/codec values - may be processing values (le...
...registration to central server based on presence of Wifi connection AND allowed time settings AND absense of call activity. 3. When SIP registration is active phone can receive SIP INVITE with a number in SIP FROM string. Upon receipt phone will initiate call to cell network to received FROM Number and then bridge audio streams between SIP and cell network call legs. 4. Application should support G711 codec (G729 only if licensing fee is not required per subscriber). We need NAT traversal mechanisms in the app. 5. While in active call bridge mode, application should display busy status, however it's important that no User interaction is necessary for application to bridge incoming call to cell network. 6. Call ends from either SIP leg or cell leg. 7. No other call flows ne...
... Main goal minimize Bandwidth in client side with quality voice . Required bandwidth compression (upto 60-80% in reality then theory) from Server X to Server Y. A usual SIP in G729 call takes 27-32kbps per port Explanation of scenario: 1. Server X (asterisk server, with static IP) receiving VoIP calls from different Carrier/Originator, with h323/sip protocol, using G711,G729 and/or G723r6.3 codec and sending calls to Server Y. 2. Server Y {Windows based Asterisk server with PRIVATE NETWORK IP, receiving calls from server X and sending to gateways (quintum, goip etc brand) or E1 cards. (Please take a look at this page for better clarity ) 3. Number of Server Y can be unlimited. 4. Number of Gateways/E1 cards per server
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I need a Java library ready to run stable IP telephony. Features: : User Agent Server. Server. Location Server. 4. RTP / RTCP and support for codec G729 Media audio and video streaming H264. 5. NAT traversal. 6. Include all listener or possible events. : example: A. public void onUserConnecting () -: here decide whether the user is authorized or not connected to sip server. b. public void onUserConnected () -: the user is connected via A. c. public void onUserMakingCall () - need to see if the user has balance and know the destination of the call. d. public void onUserDisconnectCall () - to make the end of the call billing. And other necessary events. It has to be completely customizable via java code.
...Operating System OS. C. Apart from all normal feature of the cell phone it must have the APPLICATION with built in encryption and tunnel FEATURE which will register the cell through a LOGIN and PASSWORD to a compatible Soft-switch through internet by SIM1 D. It must have a MY-SIP DIALER or such dialer that will use SIM2 to connect the desired number coming from the SoftSwitch 5. A very good voice codec must be develop which can delivery high voice quality from SIM2 ----> SIM1 ----> Softswitch -----> Origination side with less bandwidth consumption and/or capacity. Necessary test must be done to measure the bandwidth consumption is minimum which is very very important here. 5. A soft-switch preferably built in ASTERISK with mysql which will be able to do must have ...
need assistance to install g729 codec in asterisk asterisk 1.8 Freepbx need assistance to install g729 codec in asterisk asterisk 1.8 Freepbx
I need the following tasks done for Vicidial: 1. G729 Codec (opensource) installation 2. Server security Update and Brute-force Protection 3. Campaign optimization 4. 10 agent creation 5. Softphone Installation and Configuration 6. Voice file recording in mp3 format 7. Installation of GUI based file manager to download voice recording 8. Initial Training
...goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from USB flash dri...
We last had a system admin that installed OpenSIP(which can be googled to fnd out more) onto our Linux server. Therefore, 3 usernames with their corresponding passwords have been created, along with a database. It is able to make calls with these user accounts or credentials, - but when I pick up call, it mostly gets disconnected. - On the rare occasions tha...up call, it mostly gets disconnected. - On the rare occasions that the person called to could finally pick up the calls, there was no voice heard. So, absence of voice between the callers. - In addition, video call was nor even initiating. - Finally, Chat is also not working. One can check the sip confic file to realise it. - A necessary other step: To enable some codec like G722, G729, PCMA, PCMU and some video and audio ...
Android ndk and jni and java and c++ or c language , Job Description: i need programmer to create an api to be used for android project, the api functionality is for speex audio codec, audio encoding/decoding. encoding pcm wav byte data to speex encoded byte data then decoding the speex byte data back to pcm wav byte data. i have already developed push to talk android software using known ide or software called basic4android ,in order to use the codec java api, the project has to be converted to a library using very easy method.
We last had a system admin that installed OpenSIP(which can be googled to fnd out more) onto our Linux server. Therefore, 3 usernames with their corresponding passwords have been created, along with a database. It is able to make calls with these user accounts or credentials, - but when I pick up call, it mostly gets disconnected. - On the rare occasions tha...up call, it mostly gets disconnected. - On the rare occasions that the person called to could finally pick up the calls, there was no voice heard. So, absence of voice between the callers. - In addition, video call was nor even initiating. - Finally, Chat is also not working. One can check the sip confic file to realise it. - A necessary other step: To enable some codec like G722, G729, PCMA, PCMU and some video and audio ...
I need a C# wrapper for the ffmpeg codec library for converting .AVI, .WMV, .MOV, .MPEG-4, H.263 3GP, XVid MPEG-4 (AVI), Windows Media Video (V.7 WMV), .MP3, .MP4, .FLV files. In others words, I need video converter system. The wrapper will need to accept an input path for the video to be converted and an output path for the converted. ## Deliverables Complete and fully-functional working program(s) in executable form as well as complete source code of all work done.
We last had a system admin that installed OpenSIP(which can be googled to fnd out more) onto our Linux server. Therefore, 3 usernames with their corresponding passwords have been created, along with a database. It is able to make calls with these user accounts or credentials, - but when I pick up call, it mostly gets disconnected. - On the rare occasions tha...up call, it mostly gets disconnected. - On the rare occasions that the person called to could finally pick up the calls, there was no voice heard. So, absence of voice between the callers. - In addition, video call was nor even initiating. - Finally, Chat is also not working. One can check the sip confic file to realise it. - A necessary other step: To enable some codec like G722, G729, PCMA, PCMU and some video and audio ...
...from an external PHP website providing various paramets when creating the extension. We will provide server with latest asterisk pre-installed. Developer will have to program this on their own instance of asterisk, and provide code as part of deliverable. Deliverables: Just scripts below: Script 1: Creating an extension - extension number - password - server address - default codec / codec order - call waiting enabled - voicemail enabled -- voicemail to email -- voicemail attach recording -- voicemail delete vm when emailing - # of allowed channels allowable total on extension (in/out) - call forward options to external number (CID info: number is same as extenion number, display name = callers originating number) -- unavailable -- busy -- unco...
For my dissertation, I had a company create a video advertising platform. This platform played a video and then ran randomly selected advertisements (from a list I chose). The platform and experiment ran fine. I was given backup files so... for the raw database in csv format. for the sql code to recreate the database. for the video player source code. You can recompile it using flex builder or a free alternative. for the main page folder including php scripts and videos (big file). the video compression settings: Video Codec: H.264 Video Size: 640 x 480 Bit Rate: 2500 Frame rate: Original from videos. With those files / data you should be able to easily recreate the experiment.
For my dissertation, I had a company create a video advertising platform. This platform played a video and then ran randomly selected advertisements (from a list I chose). The platform and experiment ran fine. I was given backup files so... for the raw database in csv format. for the sql code to recreate the database. for the video player source code. You can recompile it using flex builder or a free alternative. for the main page folder including php scripts and videos (big file). the video compression settings: Video Codec: H.264 Video Size: 640 x 480 Bit Rate: 2500 Frame rate: Original from videos. With those files / data you should be able to easily recreate the experiment.
...for a PC software that works with our softswitch with unlimited user licence. requirements: Call Recording Call Conference video call Hold / Unhold Transfer/forward to mobile or landline DND (Do not Disturb) Redial Mute Auto accept call Codec Supported G729,G723, G711 u, G711 a, G722, GSM, iLbc, Speex/ 8000, Speex/16000 , Speex / 32000 Codec selection and Codec Quality Control (Bandwidth control) Silence Suppression Echo Cancellation Uses NEW RFC 3261 compliant stack DTMF (RFC 2833) RFC 3951: Internet Low Bit Rate Codec (iLBC) Proxy Settings Registration Timeout Acoustic Echo Cancellation. Packet concealing Packet Lost Concealment (PLC). Comfort Noise Generator (CNG), Resampling Balance Display Credit time display: C...
Hello, We are looking for a developer who develops for us ffmpeg and a player under eclipse. The player should be able to open rtmp, rtsp hardcoded links. The container consists of h264 video codec and aac audio codec. It should be kept simple and very easy, so we base on which it can be integrated into our app. I need an Android Video player (ffmpeg based) for playing live streams such as rtmp, rtsp. The project should contains the following: - One Android Activity, the player (It must have the pause, play, stop, skip time button) - ffmpeg lib (ndk) - reference: RockPlayer, MPlayer… So it can be consider as a base for a player, it only consist in one Activity, and the native C/C++ code for playing a rtmp, rtsp with ffmpeg. The build-in media player of ...
Create something familiar with : Features that needed are : Video Conference key features + many to many 2 way video chat + 3 default video panels + unlimited extra video panels + user list with online participants + room list + file sharing + private messaging + P2P group streaming support + codec setup (H264, Speex, NellyMoser) + acoustic echo cancellation + session timer and control with scripts + 100% web based for clients Developer will hand over all the files and the source code.
