Depends on what do you mean by 'network-based video game'. If you mean transfer of data such as the player's location, etc to all players then use TCP. However, if you are transmiting the screen elements in real time, then you can use UDP.
You are creating a network-based video game. What influences your decision about which transport protocol to use for the application?
The most popular example of UDP protocol usage is video streaming such as Youtube, Twitch, etc.
RTP (Real-time Transport Protocol)
The Fourth Protocol - video game - happened in 1985.
The Fourth Protocol - video game - was created in 1985.
SIP, SI Protocol, or Session Initiation Protocol is a signalling protocol used in video or voice calls, Instant Messaging (IM), file transfer and online gaming over an Internet Protocol (IP). The main function of the SIP is to create, modify or terminate multicast or unicast sessions. i.e. The SIP is what starts up the video call or online game session or whatever.
tcp is the transport control protocol and ip is the internet protocol. tcp is concerned with guaranteed delivery of packets from source to destination, while ip is the protocol used to actually deliver packets. tcp is located at layer 4 of the osi model, or the transport layer. ip is located at layer 3 of the osi model, or the network layer. check out the video tutorials in the link below.
The transport layer protocol commonly used for streaming media is User Datagram Protocol (UDP). Unlike Transmission Control Protocol (TCP), which ensures reliable delivery and order of packets, UDP allows for faster transmission by sacrificing some reliability, making it suitable for real-time applications like video and audio streaming. This is because streaming media can tolerate some data loss without significantly affecting the user experience.
RTP (Real-time Transport Protocol): Carries the voice and video traffic and doesn't do any signalling, signaling is done by H.323, MGCP or SIP protocols. Its uses UDP to carry the payload hence its not actually a protocol Where RTCP (Real-time Transport Control Protocol) Manages the quality of RTP such as QOS Identification, Session size estimation and scaling and basically works with RTP side by side.
Real-time Protocol (RTP) is primarily used for delivering audio and video over networks in real-time applications, such as video conferencing, streaming media, and VoIP (Voice over Internet Protocol). It provides end-to-end network transport functions, including payload type identification, sequence numbering, and time-stamping, which help in maintaining the quality of service. RTP is often used in conjunction with the Real-time Control Protocol (RTCP) for monitoring transmission statistics and quality.
Most streaming tv links being played in livestation are using mms protocol for asf videos file and rtmp protocol for flv videos files. For mms protocol, first open URL snooper which is a free software, then look for the links ended with asf. Then put this link into Net transport to download the live video stream. You can download Net transport by typing " last free version of net transport" in google search. Those links using rtmp protocol, you may use Replay media catcher but it is not a free one. Please beware of copying copyrighted materials.
RTP