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    1,159 java rtp znalezione projekty, cennik w USD

    Przedmiotem zlecenia jest skonfigurowanie serwera internetowego, do którego wysyłany będzie strumień audio w protokole RTP (MP3), serwer zaś ma zapewniać streaming tego materiału dla wielu odbiorców w jednym z powszechnie stosowanych systemów (np. ShoutCast, IceCast, RTSP). Założenia szczegółowe: - użycie serwera w cloudzie (np. homecloud, ovh) - serwer

    min $3
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    0 składanie ofert

    Witam,...posiadają karty TV i chciałby dodatkowo na tym zarobić. Zlecenie miałoby polegać na wysyłaniu programów z telewizji (kablowej/satelitarnej) na serwer za pomocą protokołu RTMP/RTP w minimalnej rozdzeilczości 640x480. Proponujemy rozliczenie za kanał na miesiąc. Osoby zainteresowane proszę o kontakt w celu ustalenia szczegółów.

    min $3
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    Witam, potrzebuję pomocy bądź napisania programu korzystającego z protokołu RTP. Szczegóły na maila.

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    Witam, potrzebuje bardzo prosty program w Javie: - implementacja protokołu RTSP w wersji unicast - transmisja w postaci protokołu RTP - implementacja protokołu RTSP w wersji multicast Aplikacja musi implementować dokumenty RFC a samo działania polega na przesłaniu jakiegoś pliku video z serwera dla klienta (coś jak VideoOnDemand). Program nie

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    Witam, potrzebuje bardzo prosty program w Javie: - implementacja protokołu RTSP w wersji unicast - transmisja w postaci protokołu RTP - implementacja protokołu RTSP w wersji multicast Program na zaliczenie przedmiotu na studiach, nie musi być nawet interfejsu okienkowego. Aplikacja musi implementować dokumenty RFC a samo działania polega na

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    ...być monitorowane standardowym syslogiem (zarówno po stronie klienta, jak i serwera). Adres tego serwera nie może być wkompilowany w program d) Transmisja w postaci protokołu RTP lub SRTP e) System AAA zrealizowany w oparciu o serwer RADIUS oraz jeżeli to możliwe Wykorzystanie certyfikatów PKI w systemie uwierzytelniania lub autoryzacji Z racji

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    poszukuje programisty do pomocy z integracja obslugi video streamingu dla Symbian 3rd i wyzsze. Streaming po RTP, kodek h264. Prosze o kontakt osoby posiadajace doswiadczenie w tym temacie.

    min $3
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    Witam. Przedmiotem zlecenia jest napisanie artykułu na potrzeby serwisu internetowego o tematyce transmisji głos...sieciach pakietowych tj. opoźnienia i jego rodzaje, gubienie pakietów, rozsynchronizowanie itp. - mechanizmy wprowadzajace jakość usługi tj. kolejkowanie, kompresja nagłówka RTP itp. - mozna wspomnieć o CoS i ToS. pozdrawiam

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    We need a simple video streaming application in QT on Linux with...screens. 1st Screen - Navigation Link / button (3) 2nd Screen - Video display from camera with 2 PIP (Picture in Picture) video screens to display live streaming videos (HLS or RTP) 3rd Screen - Setting screen with 3 text fields for FPS, Resolution and Bitrate Regards, Kannan R N

    $386 (Avg Bid)
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    android tv we need a iptv app with existing backend we allready have the backend providing ch...to stb's compatibility with android tv (leanback) is strictly required ability to switch the player from full screen to side by side with channels list required formats: hls, RTP, UDP (multicast) recomanded players: exo player, vitamio budget: 100 to 150USD

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    Create a simple android MIRRORING application (audio and video) to stream to PC through RTSP ,RTP or RTMP. the application will simply stream the device desktop and audio and act as a UDP/TCP streamer. Conditions: Freelancers should be living in south Korea, meet and hand the project. Payment is elementary 25$, and it can go up higher if the project

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    Need an ICE/STUN/TURN server installed in an Centos 7 server in order to have NAT We...to check and explain me how to configure Asterisk and WebRTC script (like doubango) to work when the client is behind NAT. I have coturn installed but not configurated. I see RTP packets but only in one way. Have server A with Asterisk and server B to WebRTC script.

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    I need gstreamer plugins (muxer and demuxer). The muxer inserts data via setting of property into the empty spaces of the RTP packet. For example, rtp packets are (1) 400 bytes, (2) 400 bytes, (3) 340 bytes (insert data in this packet). The demuxer does the opposite. I would need a test gstreamer application to prove test the plugins.

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    Supporting All common streaming protocols (HTTP,RTMP,RTSP,RTP,UDP,MMS) On-Demand With fast zapping time plus de détails en privé

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    ...между мобильным приложением и прибором. 2. Воспроизведение видеопотока, полученного с прибора на мобильном устройстве. Интерфейс обмена видео-данными - Wi-Fi, протокол передачи RTP/RTSP, видеоданные в формате MJPEG Необходимое оборудование будит предоставленно, по этой причине приоритет разработчикам из Украины. Готовы обсуждать сроки, стоимость, формы

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    ...supports. 14. Video overlay watermark supports. 15. Audio mp3 and aac codec supports. 16. HTTP Living Streaming with Adaptive Bitrate Streaming supports. 17. HTTP, RTMP, RTSP, RTP and UDP (and all other streaming protocols that is supported by ffmpeg) > INPUT and OUTPUT protocol supports. 18. Login password and burst force crack protection. 19. IP and

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    Olá! Estou reescrevendo uma biblioteca Javascript PURO que faz consumo de uma API HTTP e outra RTP/Websocket. A biblioteca original funciona com nosso endpoint. Estou com pouco tempo para programar e preciso de alguém que tenha disponibilidade e capacidade para terminar esta reescrita. Vou apresentar o estado atual do código e orientar os próximos

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    Hi Alex K., i have an app that convert RTP to audio file. but i need to enhance and speed the process now , i receive an rtp packet , convert to pcm and add to wav file . once all packets are received , i close the wav file , then convert to mp3. i need an algorithm to convert the rtp packet directly to an mp3 file if feasible.

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    We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.

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    ...([zaloguj się, aby wyświetlić URL]). Janus then forwards the stream in RTP over UDP to our machine (Python worker). This machine decodes the stream with Gstreamer and performs video analysis. WebApp ------------> Janus ----------------> Gstreamer WebRTC RTP Overall the architecture works, with very short latency. However

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    VOiP Tunnel solution Zakończone left

    Hello. We are looking VOIP Tunnel software solutions to bypass voip calls in blockage area . It is encrypts/decrypt SIP and RTP packets and send voip traffic between soft switches without blockage . We are looking experienced guy who already developed or worked on this .

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    dear all , i have pcap file which it capture from wire shark for my sip switch .. i want such script which will locate pcap file , and extract audio data from it .. it is RTP ... but i do not know the encode format (g711 or g722... ) the one who can have this project should first extract audio from sample i will send to him then we can start writing

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    ...on stream failure Easy & powerful Transcoding System - H.264 and H.265 support On-Demand With fast zapping time Supporting All common streaming protocols (HTTP,RTMP,RTSP,RTP,UDP,MMS) TV Archive & Timeshift MPEG-TS, HLS, RTMP Output Fingerprint Sender -Optional RTMP Push Anti-Restreaming Concurrent Connection limitation, ISP & Country Locking

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    Mandatory Skills: 1) Hands on expertise in SIP, RTP and RTCP 2) Good knowledge on TUN,STUN, NAT 3) Free-switch working knowledge on audio conference using SIP and Web RTC 4) Good knowledge on RTP Proxy and routed audio conferences concept where media would flow via free switch RTP Proxy 5) Working experience of High Availability and Cluster 6) SDP

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    Hi , This project was working perfectly on a standard ADSL modem but ive migrated to a Cisco 3945 router and it does not w...the LAN only. And sucessful in getting my 4G mobile device to work on SIP, however my issue is theres no Audio. Persumably because this relates to NVI NAT, SIP or a lack of RTP. Your job is to get this working :) Cheers Kris

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    The VMS platform project should have a universal form of support for mjpeg H264 formats.h264+. H265+ support IPv4 / IPv6, TCP, UDP, RTP, RTSP, RTCP, HTTP, HTTPS, DNS, DDNS, DHCP, FTP, NTP, SMTP, UPnP, SIP, SNMP, PPPoE, VLAN, 802.1 x, QoS, ONVIF Profile S & G, and to integrate via the SDK P2P, video Analytics modules [zaloguj się, aby wyświetlić URL]

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    We have an existing project, where we have developed in jque...plugins into react or searching for new ones. we have few bugs like - The rendering takes time in react and the js is already loaded which makes zoom to not work. we are using RTP pinch zoom jquery plugin We have used these technologies Front end - React Backend - node Api - Graphql

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    ...to the user based on tags. Content already seen by the user should not be displayed again. The Right hand side of the page links curated content - Game reviews, game data, RTP etc upvote/downvote/report No content management is need for the curate content, these page will all user the same layout template. All content can be "tagged". Users can

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    ...GeoIP Based, ISP Based ) Connection Anti-Drop on stream failure Easy & powerful Transcoding System fast zapping time Supporting All common streaming protocols (HTTP,RTMP,RTSP,RTP,UDP,MMS) TV Archive & Timeshift MPEG-TS, HLS, RTMP Output Fingerprint Sender RTMP Push Anti-Restreaming Concurrent Connection limitation, ISP & Country Locking Incredible Fast

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    Project for Rahul S. Zakończone left

    Hi Rahul S., I am looking for developer/s with experience in Voip/RTP java/c++ unix/Adnroid/iOS platforms for a digital signage app. Please contact me at ajain at [zaloguj się, aby wyświetlić URL]

    $500 (Avg Bid)
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    iPhone Developer rtp/udp stream player

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    pcap file to wav Zakończone left

    I need someone that extract the audio conversation from a pcap file. The pcap it has dynamic rtp inside, that is sure AMR/8000. I don't need any source code: i need the extracted audio file (to listen if it's ok) and an explain about how to convert the file.

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    Hi There, We require help to build a proof of concept.s We want a system connect a Client to a Carrier that is scalable. However, we do not want to disclose any of the Carrier IPs to the Client and Vice Versa. I've done some research and using OpenSIPS and RTPproxy can help with this but I am having trouble setting this up on AWS. For the proof of concept, we can use 2 PABXs one to act as a ...

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    ...communicates with Google over gRPC for Speech to Text and Amazon Poly over JSON/REST for Text to Speech. Interfaces SIP Interface - SIP signalling interface. RTP Interface - G.711 and G.722 RTP media interface. JSON/REST API - Chatbot textual interface. Speech to Text - Google gRPC streaming recognition interface. Text to Speech - Amazon Poly TTS ChatAPI

    $26 / hr (Avg Bid)
    Wyróżnienie Pilne Projekt z Umową o Zachowaniu Poufności
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    Proyecto para Panky N. Zakończone left

    ...send calls through SIP and forward them to the GSM network AND receive calls from GSM and forward to SIP. The application should then forward the audio and convert from VoIP (RTP) to GSM and vice versa. o The app should be able to run on cheap devices (+-100$). General o Deliveries o The application working in APK format o Full source code o Simple

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    ...Streams & Users (With Authentication) ✔️ Live Streams, VOD ✔️ Easy & powerful Transcoding System ✔️ On-Demand ✔️ Supporting All common streaming protocols (HTTP,RTMP,RTSP,RTP,UDP,MMS) ✔️ TV Archive & Timeshift ✔️ MPEG-TS, HLS, RTMP Output ✔️ Create your Own Live Channels by combining multiple videos ✔️ Powerful Logging, Statistics & Connection

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    I need to sync rtp with caller and called with RTCP SR. Project must be done in c# . SSRC and Payload changes. Codecs are G711,G729,AMR WB,AMR NB,EVS.

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    Hi, I need someone to get my excel list of 200 online slots games and fill out the b...someone to get my excel list of 200 online slots games and fill out the blanks with the following data for each game: Paylines Reels Progressive Software Theme Tags (max. 3) RTP Data will be verified by operators, so everything needs to be as accurate as it gets.

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    Softphone App Zakończone left

    ...License, version 2" at [zaloguj się, aby wyświetlić URL] Must be able to work with existing source primarily Java and XML and modify to specification. Familiarity with Signal and Video processing as well as SIP,XMPP,RTSP and streaming (RTP,codecs) Final result must be a working APK capable of being loaded directly to the phone without Google Play Store

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    ...under load Streaming Protocol: + HTTP + RTSP + RMTP + UDP Multicast + Re-Stream Protocol + HTTP MPEG2/H264 TS + HTTP ASF (H264) + UDP/RTP MPEG2/H264 TS Live Encoding Protocol: + UDP Multicast + RTP Multicast + RTP Unicast + UDP Unicast Player Management: + Periodically Check Players' status + Remotely Reboot Player + Remotely Configure Player Setting

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    ...****************** + HTTP + RTSP + RMTP + UDP Multicast + Re-Stream Protocol + HTTP MPEG2/H264 TS + HTTP ASF (H264) + UDP/RTP MPEG2/H264 TS Live Encoding Protocol ********************** + UDP Multicast + RTP Multicast + RTP Unicast + UDP Unicast + Recorded TV on Demand + Record TV programs to archive + Master / Slave Server + Act as

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    Need someone to be very strong with DNS setup understanding. 1. Setup account in WHM 2. Setup ALL DNS settings to point correctly. A records RTP records 3. Configure our privet name servers ns1.arcaveroom.com. 4. Make sure the hostname is setup correctly in WHM 5. After you setup the DNS i must use a DNS checker to make sure the site points correctly

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    ...****************** + HTTP + RTSP + RMTP + UDP Multicast + Re-Stream Protocol + HTTP MPEG2/H264 TS + HTTP ASF (H264) + UDP/RTP MPEG2/H264 TS Live Encoding Protocol ********************** + UDP Multicast + RTP Multicast + RTP Unicast + UDP Unicast + Recorded TV on Demand + Record TV programs to archive + Master / Slave Server + Act as

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    We are looking for a Hospitality TV App with IPTV & Services. Req.... Streaming File Types. Application should be able to play the Live Video by using an embedded VLC Player or similar player which can Play H.264, MKV, Divx, HLS, UDP, RTP, RTMP, and other available format . 12 Months Support Testing to be carried out and for 1 Month .

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    ...be the added Value . Main Future of APP. App Should be able to Play the Live Video by Using the VLC Player or Similar Player which can Play H.264, MKV, Divx, HLS, UDP, RTP, RTMP, and other available format . Additional if you can advise and use the Microsoft Live Player for HEVC Solution. So We have all those Streams available on the Server

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    Project for an1566706 Zakończone left

    ...****************** + HTTP + RTSP + RMTP + UDP Multicast + Re-Stream Protocol + HTTP MPEG2/H264 TS + HTTP ASF (H264) + UDP/RTP MPEG2/H264 TS Live Encoding Protocol ********************** + UDP Multicast + RTP Multicast + RTP Unicast + UDP Unicast + Recorded TV on Demand + Record TV programs to archive + Master / Slave Server + Act as

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    I need an x86 pentium Real time process module (extension .vxe) that would open serila port 1 and send 10 bytes of data every 100ms. You should be familiar with vxworks 6.7 or 6.8 and have your own development package to finish this simple task.

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    Java Sip API Zakończone left
    ZWERYFIKOWANY

    Hello, i need an API built for sip and rtp handling. I need it to handle things like calls and messaging. Please add “have read the description” to the beginning of your description to narrow down decisions. Thank you! :)

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    Need working Kamailio 5.1 SIP/RTP Proxy configuration. The server will be a centos and will have 2 NIC ( 1 on DMZ and 1 on LAN ) and SIP proxy must forward all SIP messages (including REGISTER, SUBSCRIBE, NOTIFY, OPTIONS, etc.) and also pass all RTP traffic through RTPENGINE to a internal Asterisk/Freepbx with TLS support. It is also important

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    Need working Kamailio 5.1 SIP/RTP Proxy configuration. The server will be a centos and will have 2 NIC ( 1 on DMZ and 1 on LAN ) and SIP proxy must forward all SIP messages (including REGISTER, SUBSCRIBE, NOTIFY, OPTIONS, etc.) and also pass all RTP traffic through RTPENGINE to a internal Asterisk/Freepbx with TLS support. It is also important

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    12 składanie ofert