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    1,159 rtp job net developer trabajados encontrados, precios en USD
    Desarrollo móvil Finalizado left

    Aplicación Android Necesito el diseño y la construcción Necesitamos desarrollar un softphone al estilo zoiper, que soporte protocolos Sip, Ssip, rtp, rtps, videollamadas, codecs 711 Y 729, H264, opux.

    $323 (Avg Bid)
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    ...y Datos, VoIP, TDM, herramientas SBC, [iniciar sesión para ver URL] de herramientas de análisis de trazas VoIP y TDM Conocimientos relacionados con los siguientes conceptos: ISUP, SS7, RTP, VOICE QUALITY, TCP/IP, REDES SIP, SBC, [iniciar sesión para ver URL] alto.-...

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    ...level</li></ul>También se valora experiencia en Amazon Web Services, message queue's (ZeroMQ, ActiveMQ, etc.), escalabilidad y protocolos de streaming de audio / video (ejemplo RTSP, RTMP, RTP, RTCP, SIP)<br /> <br /><br />Duración del proyecto: Larga duración<br />Roles a cubrir por el freelance: Desarrollador<br />...

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    Ingeniero/a SIPTrunk Finalizado left

    ...herramientas SBC, [iniciar sesión para ver URL]ón de incidencias.(Remedy)Conocimientos de herramientas de análisis de trazas VoIP y TDM Conocimientos relacionados con los siguientes conceptos: ISUP, SS7, RTP, VOICE QUALITY, TCP/IP, REDES SIP, SBC, [iniciar sesión para ver URL] valorable experiencia con equipos:- SBCs- SoftSwitch- MediaGwIngles alto.-...

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    ...log.- SCI: Configuración de alarmas, monitoreo de servicios, creación y configuración de scripts.- Conocimientos de los componenetes: SipServer, OCS, RM, MCP -- Protocolo SIP/RTP- Conocimientos de la solución eServices de GenesysCalificaciones Deseables (no excluyentes):- Conocimientos de lasherramientas SDK de Genesys y Base de Datos (SQL Server)- Experiencia

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    ...y Datos, VoIP, TDM, herramientas SBC, [iniciar sesión para ver URL] de herramientas de análisis de trazas VoIP y TDM Conocimientos relacionados con los siguientes conceptos: ISUP, SS7, RTP, VOICE QUALITY, TCP/IP, REDES SIP, SBC, [iniciar sesión para ver URL] alto.-...

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    Desarrollar software Finalizado left

    Player montado sobre una placa base Raspber...montado sobre una placa base Raspberry Pi. Con salidas de video analógico compuesto y HDMI, permite reproducir H.264 con AAC o MP3 diversos protocolos (RTMP, HLS, RTSP, HTTP, RTP, UDP, Youtube, UStream y Livestream). Latencia intrínseca de hasta 300 ms. SIMILAR AL DE ESTA PAGINA [iniciar sesión para ver URL]

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    IPTV Interactivo Finalizado left

    Queremos crear un IPTV Interactivo a través de reproductores de RTP Streaming o HTML Streaming para reproducir canales de televisión desde un servidor propio. Esos canales están guardados en una base de datos de MySQL y se cargarían en la APP vía lenguaje de programación Android o PHP, también necesitamos varias cajas que carguen WEB para imp...

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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

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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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    ...([iniciar sesión para ver 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 Finalizado 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 work ...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 [iniciar sesión para ver 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. Finalizado 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 [iniciar sesión para ver URL]

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    iPhone Developer rtp/udp stream player

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    pcap file to wav Finalizado 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

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    Destacado Urgente Acuerdo de Confidencialidad
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    Proyecto para Panky N. Finalizado 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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    I need Middle ware for IPTV service with feat...✔️ 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 Logs ✔️ Developer API

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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 Finalizado left

    ...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 and also downloadable from Google Play Store. Additional projects

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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 Finalizado 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 Finalizado left
    VERIFICADO

    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 ofertas