Rtpemplois

Night Shift IST Product: CallHippo. URL : CallHippo was launched in 2017. It is an intelligent VoIP (voice over Internet protocol) service provider for business organizations and call centers. The main objective behind launching CallHippo was to provide easy to adopt, cost-effective, scalable, and accessible phone sy...webrtc integration. - Responsible for VOIP software development. -To work on Queue, IVR and Voicemail related applications. -Responsibility of Freeswitch installation, configuration and troubleshooting. -Deployment of multiple instances of Freeswitch using a load balancer. Requirement -2+ years of experience in FreeSWITCH or other related VoIP technologies. -Good Knowledge in PBX, SIP, RTP protocols. -Experience with VOIP Software development

Hi, i need a pirate themed slot machine matching to my websites design. Every day a visitor comes, he will get 10 Freespins on that slotmachine after he signed up to our newsletter. I want the win prices to vary through a admin panel and i also would like to set the RTP%s Return to Player %s by myself if possible. I have some designs, but i need you to create the slot design based on the website design.

I need to configure Kamailo SBC on Ubuntu 18.04 to connect multiple Microsoft Teams account in the same SBC server. Require complete / usable and repeatable documentation for: - Configuration of Ka...on Ubuntu 18.04 to connect multiple Microsoft Teams account in the same SBC server. Require complete / usable and repeatable documentation for: - Configuration of Kamailio - TCP / UDP / WebRTC Proxy passing through to Asterisk on Private IP - Registration pass through to Asterisk with rewrite to present sourceip as reg contact on ast. - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk MS ...

...(using GSM codec, UDP packets), capturing audio from Android Smartphone microphone - receive, from a my server, a VoIP flow (UDP packets, always by GSM codec), playing it in Android Smartphone loudspeaker. It worked up to Android 11, but starting from Android 12 it doesn't work anymore because is deprecated. In "" no alternatives solution are mentioned, just deprecated. I need an alternative solution, so the developper should rewrite the class here uploaded with something alternative to library "" The class is self explaining and indipendent from the rest of the project code, with the exception of some methods (for example for logging) that you can ignore. Therefore, and obviously

...encoding statistics to be output to a graph in the UI for visual confirmation of the performance. b. We can start with one or two workflows with the addition of new workflows in future versions. 5. Set-up User Experience: a. UI Select: local video device (camera or capture card) b. UI Select or Enter: Encode profile (codec, bitrate, frame size, etc…) c. UI Select: Output protocol (file, rtmp, rtp, rtsp…), Enter: protocol options d. UI Enter: Destination (path, local host:port or server address:key) e. NOTE: It would be a desire to allow selection of record to file in parallel with network output. 6. Operation User Experience: a. Status: Configured/Standby b. Status: Running <- Important -> i. Display performance graphs of data from encoding pr...

Hi, We need to integrate the RTP proxy in Kamailio in the class-4/ scenario with DPDK supported network cards. After getting signal Kamailio server, the RTP proxy engine will create RTP flows using DPDK . Packets will take the bypass route, eating up less cpu cycles in the server. At present, we don't need to process any information within the RTP packets.

Hi, We need to integrate the RTP proxy in Kamailio in the class-4/ scenario with DPDK supported network cards. After getting signal Kamailio server, the RTP proxy engine will create RTP flows using DPDK . Packets will take the bypass route, eating up less cpu cycles in the server. At present, we don't need to process any information within the RTP packets. Mustafa

We need a SBC installation (Kamailio or OpenSIPS, preferably Kamailio), which will have several Asterisk servers behind, that do not use realtime (and won't use) but use a database to generate config files (both extensions and sip). Said SBC must manage the authentication, but Asterisk peers should still be independent and have their own config (things like co...their own config (things like context, call-limit, accountcode, etc.). Due to not using realtime for Asterisks, registration control should be managed in the SBC's own database. The SBC will have to load balance between the asterisk, and have failover between them. It should have the basic security modules as well (e.g. secfilter with GeoIP for Kamailio). Also it should be able to control NAT and RTP as well (rtp...

WebRTC • Experience with Strong WebRTC platforms • Experience with WebRTC (direct use and/or debugging of...and/or debugging of the WebRTC APIs) • Knowledge WebRTC server technologies. • Actively involved in all phases of our WebRTC product lifecycle. • Experience developing applications on a VoIP infrastructure • Integration of Web and Mobile Applications with WebRtc Technologies for Audio ,Video , Screen Sharing and Meeting • Knowledge and experience of the following protocols and platforms: SIP, WebRTC, RTP, REST APIs, Linux • Knowledge and experience with one or more of the following: Asterisk, OpenSIPS, Kamailio • Good knowledge of media codecs, formats, transports and container protocol Webrtc • Good knowledge on Video Transcod...

Implement an H265 packetizer and depacketizer according to the H265 specification in the open source library Pion: A reference implementation can be found in gstreamer The pion one should function identically

Need to take multiple WAN live video sources and repackage without transcoding to rtp on a specific network card...solution needs to be repeatable and have interruption safety nets. Preferably a solution that can ingest a file to ease administration or changes.

I'm looking for someone help me edit/develop node...looking for someone help me edit/develop nodejs function about munipulate the roulette result in my nodejs app. Now it's random result but I want it to hold 5% from total rollover and current game rollover like the picture attached. . So after this success we will always hold the 5% profitable from total rollover. Please see the attached image and the example detailed below to understand it. Example RTP Calculation: -> If total profit is 50000 (over 5%), and all game rollover and the current game is only 100,000 the result will be random normally. -> If total profit is only 4500 (<5%), and all game rollover and current game is 100000 we will need to munipulate the result to hold the profitable side. For more i...

I need support to create a raw code (an ionic class) for Cordova/Ionic that can connect to an RTSP stream (such as rtsp://: ) and it can render the frames to an img tag or a div tag. Currently, I am using and cordova-plugin-chrome-apps-sockets-udp, I can create a connection with RTSP and RTP socket getting some raw data, but I can't convert it into an image.

...*************** All with VLC and FFSERVER Video / Audio Format ********************* + MPEG2/MPEG-AVC/H264 + AAC/MP3 Containers ********** + MP4, Transport Stream, MOV, 3GP, 3G2 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 + Recorded TV on Demand + Record TV programs to archive + Master / Slave Server + Act as a Master Server, a Slave server or both + HTTP protocol between Master and Slave Server Player Management ***************** + Periodically Check Players' status + Remotel...

H264 Videostream (RTP) to be read on specific Port on Windows PC and fed to Milestone VMS for recording and display. Custom Text and Icon overlay to be added to this incoming video stream.

RTP and PSSA Header and footer and Enfimye Background Fixation

Summary We specialize in the transport of video media from inhospitable locations, primarily in service of the global news industry, and is seeking a senior-level network engineer with expertise in implementing and optimizing minimal-delay video streaming solutions using technologies such as RIST, SRT, RTSP, and RTP, including transmission over peer-to-peer connections. Experience using Janus WebRTC Server, GStreamer, and FFMPEG is also a requirement. The ideal candidate would have reasonable comfortability in processing streams from capture devices and converting encoded streams for playout. Ideally, this candidate would be familiar with transport protocols and may be asked to implement custom transport protocols or tune existing protocols. Familiarity with third-party streaming s...

i need experienced RTP and PROXY specialist

M. WAYNE MCNEILL PROJECT November 2021 Greetings, I have a large job and a small, fixed budget. I am in dire need of an interactive 3D rendering of a vision God has laid on me for a client in the RTP area of North Carolina. I understand if you rather not accept this assignment, but will you please be up front with me as to whether we will be able to work together on this project? Until a deal is consummated, I can pay $450.00 for this work. I have a vision for a 47-acre tract of land for my customer. I know that if my client sees and likes what I have pictured in my head and you can capture on canvass, there will be many more projects to come. I have faith that I will find the perfect person or outfit to turn my vision into a reality. Allow me to start with the entrance to the p...

Need to develop Real-time transport protocol (RTP) application or to find an existing one

M. WAYNE MCNEILL PROJECT November 2021 Greetings, I have a large job and a small, fixed budget. I am in dire need of an interactive 3D rendering of a vision God has laid on me for a client in the RTP area of North Carolina. I understand if you rather not accept this assignment, but will you please be up front with me as to whether we will be able to work together on this project? Until a deal is consummated, I can pay $450.00 for this work. I have a vision for a 47-acre tract of land for my customer. I know that if my client sees and likes what I have pictured in my head and you can capture on canvass, there will be many more projects to come. I have faith that I will find the perfect person or outfit to turn my vision into a reality. Allow me to start with the entrance to the p...

...input1 is down auto connect second backup stream + overview show all streams, posiblity to sort by highest uptime downtime, show stream information like bitrate, audio codec, video codec , transcoding profile (h265 / h264) fps, stop stream restart stream edit stream buttons, delete stream - nginx management + generate ouput streams MPEG-TS, HLS, RTMP, m3u8 + accept input streams HTTP,RTMP,RTSP,RTP, + nginx rtmp port change able + nginx hls streaming port change able + restart nginx all streams from jetsons should be rtmp it will be added in main system and from main system channels will be streamed outside. - cluster manager + create add new jetson kits by entering clustername , host ip, ssh port, root and passwd, network device id selection for showing realtime trougput ens...

Busco un programador que me cree una pag web con un tracker, que guarde informacion de tiradas, ganancias, volatilidad y rtp de las slots (tragamonedas) de casino online, el precio lo pone usted.

Need someone who know how to Code a User Expereince in Slot gaming. All complex Win and Lose so that match a good played game User will lose and win not just static working on RTP. The RTP will be match in the end on some percentage that i can choose. Please contact and add me on Skype Skype: asat1962

... (TLS ... etc ) GUARD SBC will get all RTP to the related IP/PBX according to USER. IP/PBX Won' t be accessible to PUBLIC and will be covered for Frauds. GUARD SBC will be flexible and capable and eligable to get register all SIP Phone Applications ( Zoiper, LinPhone, Wave Lite... etc... ) GUARD SBC will be unbreakable to well-known attacks. GUARD SBC could be DOCKERIZED to have capability to fast start, load balance, backup senarios. Basicly this GUARD SBC will be the PUBLIC face for USER to get register from all overthe world for IP/PBX but we will keep safe IP/PBX at background for Fraud and other security problems. We need to now what is the capacity for each GUARD SBC server for Concurrent CALL. When TWO USER calls each other, We want call RTP go over from IP/P...

...Deposit via QR CODE Payment Gateway (I will provide the information.) with history. - Withdrawal function and history for users. (Also withdrawal manager for admin.) - Betting History for users. (Also all game history for admin with filter.) - Show live betting pool. (with some functionally) - Admin panel (deposit/withdraw manager, user manager, system config, game config). - The system is Fair game. (RTP will caculating from rollover). - Those support via tickets or live chat. - Musical arrangement of rounds. - Jackpot function for roulette like csgo500. (Calculating from total profits) - Bot features The script is written in PHP, JS, MySQL We will judge all the details during the creation process. What would you like to get in the end: * Beautiful, functional website with mo...

I want to hire someone who can develop Xtream UI IPTV App which will support RTP and UDP live channel. I have example of the App to follow and all the graphic design ready for the job

...the following project. Support for EVS audio codec 1 Introduction A .net DLL and/or Com DLL to support decoding of recorded audio call with EVS codec. The idea is to build a .net DLL and/or Com DLL which can run on Windows (x64) platform and capable of decoding recorded media using the EVS codec in which it was recorded. 2. Key information PCAP file has all RTP/SIP for a call and will not contain any other packets. Only SIP and RTP are related to one call. Sampling rates- 8, 16, 32, and 48 kHz Modes & Encoded Bandwidth- NB (20-4000 Hz) WB (20-8000 Hz) SWB (20-16000 Hz) FB (20-20000 Hz) 3. Requirements o The end software should be a .net DLL and/or Com DLL. o C++/C# Source code of the dll created. o It should Expose a few methods/functions to use. o The main decr...

I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Opensips - Install Opensips-cli - Install Opensips-cp - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk

...the following project. Support for EVS audio codec 1 Introduction A .net DLL and/or Com DLL to support decoding of recorded audio call with EVS codec. The idea is to build a .net DLL and/or Com DLL which can run on Windows (x64) platform and capable of decoding recorded media using the EVS codec in which it was recorded. 2. Key information PCAP file has all RTP/SIP for a call and will not contain any other packets. Only SIP and RTP are related to one call. Sampling rates- 8, 16, 32, and 48 kHz Modes & Encoded Bandwidth- NB (20-4000 Hz) WB (20-8000 Hz) SWB (20-16000 Hz) FB (20-20000 Hz) 3. Requirements o The end software should be a .net DLL and/or Com DLL. o C++/C# Source code of the dll created. o It should Expose a few methods/functions to use. o The main decr...

goautodial using ASTPP trunk will having problem for autodial. customer answered the call but get silence and it wont pass to agent softphone. within 5 second will drop the call. tested using manual call with no problem with RTP voice complete. i has been try to use other supplier trunk and getting not issue with manual and auto dial from campaign.

● I am looking for a senior developer or a small team of senior developers that can develop a program as follows: - Cross-platform program for Windows and Android. - Languages: C#, C++ and Java(Android). - IDE: Visual Studio, Android Studio. ● Qualification for applying: - Extensive experience with RTP/RTSP protocols. - Knowledge of GStreamer - Understanding of getting input data from USB external devices - such as the HID class, HidP_GetCaps, etc. - Languages: C#, Android (Java) - professional level, senior developer. 　　C++ - good understanding ● ### Notes ### - This is NOT a web app. If specialized in Web technologies, no need to apply. - Do not bid if you know you don't match all conditions required. - Automatic bids will be dismissed.

...Cell Measurements, HSPA+ DC in use, Audio Codec Type, AMR Bitrate must be captured, decoded, encoded, displayed. - 2G/GSM: Band, RxLev, RxQual, C/I, TxPower, ARFCN, RRC State,Serving Cell measurements, Neighbor Cell Measurements, Audio Codec Type, AMR Bitrate ... - 4G/LTE: Band, RSRP, SINR, RSRQ, RSSI, EARFCN, RRC State ... - 5G NR: Band, SS-RSRP, SS-RSRQ, SS-SINR, BeamIndex, SS-PCI - VoLTE/RTP/SIP Layer - Lock NV Item Chipset based api - Layer-3 Messages - GSM RR, WCDMA/LTE/5G RRC/NAS, CC, MM, SIP Volte IMS, ... - log mask features - start stop script - Display on map/indoor map - License module - Upload logfiles - script editing locally and in cloud The program must run on Qualcomm based phones. It must be tested and run on sd855 chipset Requirements for partici...

...understanding of network protocols and associated setup required. Skills required: • Hands on experience in JavaScript, Java, C++ etc. • Extensive technical knowledge and working experience in Data transport services, technologies. • Work experience and knowledge in SIP based telephony solution (VoIP). • Working experience on Datacenter, Cloud environment. • Deep knowledge of protocols such as SIP, RTP/SRTP including TLS, webRTC • Very good knowledge of protocol and packet analysis • Experience working with switching and routing protocols • Strong understanding of NAT traversal for VOIP (incl. STUN/TURN) • Experience working with Linux and Windows server operating systems • Knowledge of third party/open source SIP stacks • S...

...recording, and cue/review-while-recording of H264/5 RTP video streams from various sources. We originally outsourced all video-related development to a developer who has become unavailable. We would like to be able to continue to focus on other aspects of our application without taking time to become video streaming experts for troubleshooting and further development. At the moment, we're having difficulties working with a particular camera video source that our client will be using: Depending on settings, live video displays at half rate of incoming video stream (thus accumulating lag), or live video displays with only intermittent lag but recorded video displays at half rate. The camera has options to provide "RTP/AVP", "MP2TS over RTP"...

Hi I've got several slot machine, which have been installed Ubuntu OS. I want to have some slot game embedded into my machines. There are many candidates who can develop web online slot game, but unfortunately, rarely who can do this embedded slot game development. My previous developer selected Qt desktop for ubuntu OS, but he disappeared due to his private. I hope someone help me with this project to the end. All the core should be there on the server, while on the slot machine, there should be only UI and graphics. There is internet connection between the server and the slot machines. This project is fixed contract. My maximum budget $800. For the server Live play video If this slot machine would be successful, I would be willing to move to the online slot business here. Th...

Hello, we need build audiorecognition system to detect ads in realtime from live stream ie radio from udp, rtp, http it must running under linux debian 10 it must be in c so it will be fast As PROOF YOU READED IT: write why do u think you are really skilled on realtime audiorecognition. Only for really advanced users. Many thanks!

i want to get high quality graphic for slot machine game. that need the function of free spin and adjusting RTP (return to player "winning rate"). that should support for android and ios and web

I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Opensips - Install Opensips-cli - Install Opensips-cp - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk

We need a flyer made to advertise our event for Aged Care residents. We are hosting a Morning Tea on Friday May 28th at 11am. Text is: We would like to welcome all residents to an information session about the new technology service provider, RTP Solutions. We invite you to come and learn about our products and pricing from your new service provider at 11am, Friday 28th May, in the lower level of the clubhouse. RTP Solutions is powered by Managed 247 Technology Solutions Pty. Limited

Hi, are you aware how to program the RGN for the Slot game? I developed the simple Slot game, but I used the default RNG for that. I'd like to have my own RNG to adjust RTP actively. If you think suitable for this, feel free to apply.

Hi, are you aware how to program the RGN for the Slot game? I developed the simple Slot game, but I used the default RNG for that. I'd like to have my own RNG to adjust RTP actively. If you think suitable for this, feel free to apply.

Hi I have a slots game written in pixijs, it works well. But the game logic is buggy and set winrate/RTP doesnt work. Its an Amatic card game based on Goldsvet Laravel platform if you know that brand.

As part of the Content Unit which creates and design online games for the gambling industry, we need a game mathematician / simulation developer to validate our mathematical model. We have a math model and PAR sheet available in Excel, and need it to be validated (RTP) so that around 5 billion simulations are run with these results. Minimum: • Offer detailed analysis and report of mathematical model provided • Suggest adjustments • Validate, simulate (with the help of simulation developers) and sign off mathematical models to ensure they work as intended Preferred: • Understand successful elements of slot games • Develop and explore innovative functionality of slot games • Contribute original ideas to game logic and game design team Qualifications ...

...actions of the server. The server will use the real-time protocol (RTP) to packetize the video for transport over UDP. You may reuse the code that partially implements RTSP and RTP at the client and server (link provided in the references section). When you are finished, you will have created a client-server application that does the following: - The client can pick the video file to consume using the video file name. - The client can send SETUP, PLAY, PAUSE, and TEARDOWN RTSP commands to server - The server responds to the commands above. - When the server is in the playing state, it periodically grabs a stored JPEG frame, packetizes the frame with RTP, and sends the RTP packet into a UDP socket. - The client receives the RTP packets, removes the ...

I need to configure asterisk to hide or mask the RTP IP of the vendor by masking the IP with a name. Please let me know if you have experience in this.

I want to HA-Setup of Kamailo SBC to connect multiple Microsoft Teams account. So in this Project: * Install Kamailo-Cluster * Install Opensips (with CLI and CP) * Configure TLS Certificates * Configuration RTP Proxy / RTP Engine * Setup Inbound / Outbound Routing * Setup Security best-practice Example MS Teams Tenant A <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk) MS Teams Tenant B <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk)

I want to HA-Setup of Kamailo SBC to connect multiple Microsoft Teams account. So in this Project: * Install Kamailo-Cluster * Install Opensips (with CLI and CP) * Configure TLS Certificates * Configuration RTP Proxy / RTP Engine * Setup Inbound / Outbound Routing * Setup Security best-practice Example MS Teams Tenant A <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk) MS Teams Tenant B <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk)

Requirements: - call reject/hang up, - call transfer, - no activity detection/silence detection, - DTMF detection, - Linux platform kernel version from 2.6.

Please let me know if you have experience in above tools. I've some little things to do. First project is. I need to mask the RTP IP of the originating site with any IP which is not present in the server. So the termination site will see a different RTP IP than the original one. More details will be discussed on chat.