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    205 java speex jobb har hittats, med prissättning USD

    Make PC to phone or PC to PC calls via VoIP providers Multi User Confer...to the phone line Play text using Text-To-Speech (TTS) voices Send DTMF (Inbound, SIP INFO, RFC2833) Hold calls Select Audio In/Out Devices G.711 A-law/Mu-law GSM.610 Speex Handle multiple SIP calls at the same time i want update [Logga in för att visa URL]

    $1250 (Avg Bid)
    Framhävt
    $1250 Snittbud
    2 bud
    Voip Softphone - 2 Avslutades left

    Make PC to phone or PC to PC calls via VoIP providers Multi User Confer...to the phone line Play text using Text-To-Speech (TTS) voices Send DTMF (Inbound, SIP INFO, RFC2833) Hold calls Select Audio In/Out Devices G.711 A-law/Mu-law GSM.610 Speex Handle multiple SIP calls at the same time i want update [Logga in för att visa URL]

    $2636 (Avg Bid)
    Framhävt
    $2636 Snittbud
    2 bud
    Voip Softphone Avslutades left

    Make PC to phone or PC to PC calls via VoIP providers Multi User Confer...to the phone line Play text using Text-To-Speech (TTS) voices Send DTMF (Inbound, SIP INFO, RFC2833) Hold calls Select Audio In/Out Devices G.711 A-law/Mu-law GSM.610 Speex Handle multiple SIP calls at the same time i want update [Logga in för att visa URL]

    $750 - $1500
    Framhävt
    $750 - $1500
    0 bud

    I am looking for an SDK devel...data streaming – SIP On the Apps side we need the data stream protocol. for video, there is H263, H264, VP8. for the voice, there is PCMA, Speex, GSM. We will need the APIs for IOS (Object C) and Android (java) Is this something you can help with or do you know somebody that can. I await your reply… Rgds

    $1483 (Avg Bid)
    $1483 Snittbud
    13 bud

    ...choice. 2. Encode those samples using the freely available Speex codec ([Logga in för att visa URL]). A library precompiled for iOS is available here: [Logga in för att visa URL] This is the first stage of a project whose end result is to be able to feed the resulting Speex stream to an Adobe Flash/AIR client for playback. This is

    $422 (Avg Bid)
    $422 Snittbud
    9 bud

    ...looking for a good open source project for my IAX application with good voice quality and couldn't found a good sample project for making and receiving calls with IAXClients and Speex Codec in Objective C. Note : For Bidders, Only bid if you have worked on it previously and can provide a demo for testing before awarding the project. I have downloaded

    $1681 (Avg Bid)
    $1681 Snittbud
    11 bud

    I need to Install the Speex-fec codec in Asterisk or make asterisk believe its install Note I need asterisk to present Speex-fec as an option codec even if it doesn't work

    $202 (Avg Bid)
    $202 Snittbud
    4 bud

    ...android, Iphone and windows with the all the basic futures like call conf. video calling, call forwarding etc. it must support all the codes like alaw, u law, gsm, g729,g722,h264,speex etc. pls send the details of ur product , if anything already developed which suites my requirement. It should work on all the popular smart phones available. Must have crystal

    $519 (Avg Bid)
    $519 Snittbud
    5 bud
    Voip. Sip Phone Avslutades left

    ...android, Iphone and windows with the all the basic futures like call conf. video calling, call forwarding etc. it must support all the codes like alaw, u law, gsm, g729,g722,h264,speex etc. pls send the details of ur product , if anything already developed which suites my requirement. It should work on all the popular smart phones available. Must have crystal

    $410 - $410
    $410 - $410
    0 bud

    I am looking for a good open source project for my IAX application and couldn't found a good sample project for making and receiving calls with IAXClients and Speex Codec in Objective C. I have downloaded [Logga in för att visa URL] an Open source project which was uploaded 4 years ago. I tried to make it compile and work with IAX protocols and

    $659 (Avg Bid)
    $659 Snittbud
    4 bud

    ...thing in private voice chat! Also if there are more then 35 users then to some users voice doesnt relays. I want you to fix voice delays in 2 audio settings nellymoser and speex and create 3rd audio settings named as gyaro which uses audio settings which i will be asking. I will provide all source code. Users online but hidden : sometimes this

    $178 (Avg Bid)
    $178 Snittbud
    2 bud

    ...thing in private voice chat! Also if there are more then 35 users then to some users voice doesnt relays. I want you to fix voice delays in 2 audio settings nellymoser and speex and create 3rd audio settings named as gyaro which uses audio settings which i will be asking. I will provide all source code. Users online but hidden : sometimes this

    $75 (Avg Bid)
    $75 Snittbud
    1 bud

    ...the bytes to Speex ([Logga in för att visa URL]) Write the bytes (of Speex) to the webservice body in chunks (instead of the wav file contents) On stop, stop writing on the web service and send the final chunk (zero-length). Wait for JSON results. References: [Logga in för att visa URL] [Logga in för att visa URL] http://en

    $457 (Avg Bid)
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    4 bud

    ...display (in admin room list) bandwidth Mb/min per stream for each room based on streaming settings (maxCamBandwidth in bytes/s) + sound settings [Logga in för att visa URL] (from kbps, divide by 8 to get kb/s) - room setting: Maximum Bandwidth Capacity: 10 (Mb/s) + Calculate and display maximum room capacity based on bandwidth

    PHP
    $257 (Avg Bid)
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    1 bud

    ...display (in admin room list) bandwidth Mb/min per stream for each room based on streaming settings (maxCamBandwidth in bytes/s) + sound settings [Logga in för att visa URL] (from kbps, divide by 8 to get kb/s) - room setting: Maximum Bandwidth Capacity: 10 (Mb/s) + Calculate and display maximum room capacity based on bandwidth

    PHP
    $570 (Avg Bid)
    $570 Snittbud
    4 bud

    ...(in admin room list) bandwidth per minute per stream for each room based on streaming settings (maxCamBandwidth in bytes/s) + sound settings [Logga in för att visa URL] (kbps, divide by 8 to get kb/s) - room setting: Maximum Bandwidth Capacity: 10 (Mb/s) + Calculate and display maximum room capacity based on bandwidth. Each

    PHP
    $300 (Avg Bid)
    $300 Snittbud
    1 bud

    ...display (in admin room list) bandwidth Mb/min per stream for each room based on streaming settings (maxCamBandwidth in bytes/s) + sound settings [Logga in för att visa URL] (from kbps, divide by 8 to get kb/s) - room setting: Maximum Bandwidth Capacity: 10 (Mb/s) + Calculate and display maximum room capacity based on bandwidth

    PHP
    $989 (Avg Bid)
    $989 Snittbud
    5 bud

    ...display (in admin room list) bandwidth Mb/min per stream for each room based on streaming settings (maxCamBandwidth in bytes/s) + sound settings [Logga in för att visa URL] (from kbps, divide by 8 to get kb/s) - room setting: Maximum Bandwidth Capacity: 10 (Mb/s) + Calculate and display maximum room capacity based on bandwidth

    PHP
    $812 (Avg Bid)
    $812 Snittbud
    7 bud

    Hi,...delay issues and when there are more then 30 users the chat slows down, its maybe because of chat history or private voice chats. There are 2 audio codes nellymoser and speex which you need to work on, additionally you can create new audio codec. This project uses php/mysql/red5 media server. I am online for quick questions and answers.

    $150 (Avg Bid)
    $150 Snittbud
    2 bud

    ...Skype Calls • • Supports Popular SIP /VoIP Providers • • SIP Trunking Support • • Free Communication Links to 3CX and other SIP Servers • G711 (a law and u law), GSM, Speex, IIbc • • G722 • • G729* • Advanced Real time Queue Statistics Monitor queue status Review the number of callers in a queue Log agents in and ...

    $25000 (Avg Bid)
    Framhävt Brådskande
    $25000 Snittbud
    1 bud

    ...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

    $781 (Avg Bid)
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    8 bud

    SIP Mobile Dialer (iPhone, Android and PC) with VPN Tunnel: It Sh...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 Private IP address

    $786 (Avg Bid)
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    6 bud

    ...environment). The application uses Asterisk to manage the SIP layer, and the open source framework Doubango. We need a very high quality video call, and we have chosen codecs speex and h264. The application also handles the delivery of audio or text messages in case of no response from one of the terminal. The application is around 75% complete, so

    $1720 (Avg Bid)
    $1720 Snittbud
    9 bud

    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

    $60 (Avg Bid)
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    1 bud

    ...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

    $687 (Avg Bid)
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    4 bud
    Live Video Conference Avslutades left

    ...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

    $1268 (Avg Bid)
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    8 bud

    ...it can take the code Design of program is clean and elegant, I have already QT UI protocol SIP Functions: contacts list call softphone to softphone using codec celt, speex. .... call to outbound: codec g711 - 729 Conference call to 3-4 users Video call Chating File transfer Presence user Avatar History of calls SMS Prepayment after

    $1536 (Avg Bid)
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    10 bud
    Voip System Avslutades left

    ...LCR, BRS, load balancing, failovering) - Class5 features (PBX features, ACD, call forward, transfer, DID, etc) - Codec support: all (G.729, G.723, G.711, G.722, G726, GSM, speex, iLBC, wideband, ultra-wideband, H264, DTMF, etc) - Video, Presence, IM support (SIP SIMPLE) IVR Basic calling cards (requires IVR basic) - P2P, callback (requires IVR basic)

    $9071 (Avg Bid)
    Framhävt
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    7 bud

    i need PAID programmer to provide me with working java class and source code that includes api or functions that uses JSPEEX Codec to Encode/Decode Pcm wav Byte array to speex format , i have an android application that captured microphone pcm wav data live from AudioRecord and feed it to tcp socket and send it to an EchoServer wich then get echoed

    $100 - $250
    $100 - $250
    0 bud

    ...bar 5- remove all unwanted config options in the dialer 6- sip server address hard coded. 7- CSip dialer with gingerbread theme. 8- g.729 codecs (main) GSM, PCMu, PCMa, speex, codecs. Asterisk a2billing integration 1- show user balance from Asterisk a2billing on the main screen. 2- new user registration integrated with a2billing 3- login

    $1316 (Avg Bid)
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    6 bud

    i need PAID programmer to provide me with working java class and source code that includes api or functions that uses JSPEEX Codec to Encode/Decode Pcm wav Byte array to speex format , i have an android application that captured microphone pcm wav data live from AudioRecord and feed it to tcp socket and send it to an EchoServer wich then get echoed

    $190 (Avg Bid)
    $190 Snittbud
    2 bud

    ...bar remove all unwanted config options in the dialer sip server address hardcoded. CSip dialer with gingerbread theme. g.729 codecs (main) GSM, PCMu, PCMa, speex, codecs. a2billing integration show user balance from Asterisk a2billing on the main screen. new user registration integrated with a2billing login with

    $30 - $250
    $30 - $250
    0 bud

    ...with working java class and source code that uses JSPEEX Codec to Encode/Decode Pcm wav Byte array to speex format , i have an android application that captured microphone pcm wav data live from AudioRecord and feed it to tcp socket and send it to an EchoServer wich then get echoed back to the android application to get Decoded from speex format to PCM

    $250 (Avg Bid)
    $250 Snittbud
    1 bud

    ...Connection: GPRS, EDGE, WiFi, 3G, 4G (all above 15 kbits) # Built-in tunneling and encryption (for both the signaling and the media) # Codec: G.711 (PCMU and PCMA), GSM, Speex, iLBC, G.729 G.723 # Automatic fine-tuning (including codec selection) depending on your peer capabilities, network speed and CPU power # AEC (acoustic echo canceller), Denoise

    $3268 (Avg Bid)
    $3268 Snittbud
    8 bud

    ...Connection: GPRS, EDGE, WiFi, 3G, 4G (all above 15 kbits) # Built-in tunneling and encryption (for both the signaling and the media) # Codec: G.711 (PCMU and PCMA), GSM, Speex, iLBC, G.729 G.723 # Automatic fine-tuning (including codec selection) depending on your peer capabilities, network speed and CPU power # AEC (acoustic echo canceller), Denoise

    $1080 (Avg Bid)
    $1080 Snittbud
    5 bud

    ...loads in the client web browser via the Adobe Flash Player. We need a gateway to integrate our audio teleconference bridge with our current web meeting VoIP stream. We use the Speex codec for the audio compression. We have found [Logga in för att visa URL] to offer the gateway needed for this integration but are looking for someone with recent experience working

    $30 - $150
    $30 - $150
    0 bud

    ...loads in the client web browser via the Adobe Flash Player. We need a gateway to integrate our audio teleconference bridge with our current web meeting VoIP stream. We use the Speex codec for the audio compression. We have found [Logga in för att visa URL] to offer the gateway needed for this integration but are looking for someone with recent experience working

    PHP
    $100 - $400
    $100 - $400
    0 bud

    ...and the audio has been sounding perfect. We are upgrading Videowhisper to a newer version and they now offer 2 different audio codecs to choose from, speex and nellymoser. We have tested Speex and so far it has been terrible with MP3 audio. So we figured we would use the old nellymoser audio codec combined with the new H264 video codec. Here

    $200 (Avg Bid)
    $200 Snittbud
    1 bud

    ...Connection: GPRS, EDGE, WiFi, 3G, 4G (all above 15 kbits) # Built-in tunneling and encryption (for both the signaling and the media) # Codec: G.711 (PCMU and PCMA), GSM, Speex, iLBC, G.729 G.723 # Automatic fine-tuning (including codec selection) depending on your peer capabilities, network speed and CPU power # AEC (acoustic echo canceller), Denoise

    $1500 (Avg Bid)
    $1500 Snittbud
    3 bud

    Hello, i have a simple flash application (videochat) divided in 2 parts: the first streams video from webcam to a server (red5), the second shows the video. ...working actionscript (hundreds lines of code). I need to rewrite (change) both parts from current Flash 8 + AS2 to Flash 10+AS3. Main reason of rewriting is ability to switch to Speex codec.

    $173 (Avg Bid)
    $173 Snittbud
    10 bud

    ...EDGE and 3G. The requirement are as follows: 1. Mobile P2P Voip with SIP - Sip with GSM phone number - Sip to Sip call over GPRS, EDGE and 3G - Video Call - using speex codec - To allowed subscriber choose to call using GSM or SIP 2. Messenger - Messenger to be integrated with our social network messenger - Messenger to SMS - Emotion

    $25100 (Avg Bid)
    Brådskande
    $25100 Snittbud
    15 bud

    ...Connection: GPRS, EDGE, WiFi, 3G, 4G (all above 15 kbits) # Built-in tunneling and encryption (for both the signaling and the media) # Codec: G.711 (PCMU and PCMA), GSM, Speex, iLBC, G.729 G.723 # Automatic fine-tuning (including codec selection) depending on your peer capabilities, network speed and CPU power # AEC (acoustic echo canceller), Denoise

    $1390 (Avg Bid)
    $1390 Snittbud
    11 bud

    ...want a simple Android and/or Symbian app that can do the following: 1. Record sound from the mic and save to a file with Speex codec in real-time, i.e the file should be ready when the recording stops. 2. Upload the SPEEX file to a server using HTTP POST 3. Receive JSON back from the server 4. Read data from the JSON and do another HTTP POST

    $573 (Avg Bid)
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    11 bud

    ...is not required, strictly ULAW and/or ALAW uncompressed audio, which can be sent from HTML 5. Special bonus available for implementing audio compression, such as g729 or SpeeX. *** Must run on -ANY- HTML 5 compatible browser that supports Web Sockets. *** Must register (user/pass authentication) via TCP or UDP with SIP proxy, programmers choice

    $5836 (Avg Bid)
    $5836 Snittbud
    15 bud
    Iphone VOIP/ Sip App Avslutades left

    ... We require a soft-phone for iPhone and android. Few of the details are as below Platform: iPhone (later in year Android option) Audio codecs: g729 , ulaw , alwav , speex , gsm Should be able to support STUN. Connection over Wifi / 3G / Local Access • Native Contact Phonebook • Dial, Hang up, Redial automatically • Call Hold • Call Mute

    $1283 (Avg Bid)
    $1283 Snittbud
    6 bud
    Audio Codec Decoder Avslutades left

    ...whos calling whom. Also we are able to extract the RAW Voice data. Now this RAW data could be using any codec. Some of the standard codecs like u-law, a-law, [Logga in för att visa URL] series, speex etc we are able to decode using certain code and tools. The trouble is with certain otehr codecs specifically the ISAC codec. The ISAC codec was earlier proprietary but the

    $650 (Avg Bid)
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    1 bud

    Dear , We have a running asterisk to asterisk calls service. we run now VPN on both server with openvpn and send calls from one server to another with IAX2 and codec Speex. Calls running fine but, we face calls ACD issue . 90% calls drop in 1 minute. while we dial manually its working fine but, while we send live calls most of calls connect and disconnect

    $233 (Avg Bid)
    $233 Snittbud
    6 bud
    android dialer Avslutades left

    Making and receiving SIP calls Video calling using H264 and H263+ codecs Multitasking, receiving calls in backgroun...Destination, rate information Time, call status logs Last calls log Redial Hold Sending and receiving SMS Echo cancellation Supported codecs: Audio G729, g722, G711, Speex (16 KHz), iLBC and GSM. Sign up link Recharge link

    $708 (Avg Bid)
    $708 Snittbud
    6 bud

    Making and receiving SIP calls Multitasking, receiving calls in background and sleep mode (using TCP connection) Mult...Destination, rate information Time, call status logs Last calls log Redial Hold Sending and receiving SMS Echo cancellation Supported codecs: Audio G729, g722, G711, Speex (16 KHz), iLBC and GSM. Sign up link Recharge link

    $550 (Avg Bid)
    $550 Snittbud
    1 bud

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