Add UI Elements to Unity Netcode Multiplayer Game

25.0 USD

25.0 USD peopleperhour 技术与编程 海外
686天前

详细信息

Hi there,
We have a Multiplayer Penalty Shootout Unity game created in NetCode.(check the attached gameplay sample)
The gameplay is pretty simple. It's a penalty shootout with 2 players.
- Each player takes 3 shoots, and another player acts as a goalkeeper.- The player & Goalkeeper can pick out of 5 options to move the ball & to jump.- IF both player & goalkeeper picks the same number(let's say 3), then goalkeeper saves the ball.
All lobby, and gameplay functionality are ready.
We need the following UI elements:
1. We added Goalkeeper & Player 3D character(.fbx files) & animations(maximo.com) to our game:
- So the current issue is, the gameplay elements(player, goalkeeper and etc.) are only present in the Host server. We need to host it on both the host & client side.
1. Player script -> it's written, but only for the Host side. 2. Goalkeeper script -> you need to write it for both host-client side.
Warning: It's not in Photon Engine, So you need to know how to do it on NetCode.
2. Add ScoreBoard UI Element(check attached scoreboard.png) - I will give you all elements.
3. Countdown timer for each round (7 seconds)(functionality is ready)
- So in each round, players are given 7 seconds to pick options to move the ball. You need to add a simple countdown timer from 7.
----
All logic, gameplay, winner decision logic & functionalities & 3D models & animations(on maximo.com) are ready, all you need to do is apply UI elements for both sides and UI reskin.
If you can do it today, I can pay $25 for it.
Thanks.

免责声明

该外包需求信息来源于站外平台,本站仅提供公开信息部分字段展示与订阅服务,更多请查看免责声明

关注公众号,不定期副业成功案例分享
关注公众号

不定期副业成功案例分享

领先一步获取最新的外包任务吗?

立即订阅

类似推荐

We need assistance from an experienced VoIP professional to configure call forwarding settings for our SIP trunks. Our business utilizes an Asterisk-based phone system connected to multiple SIP providers for outgoing and incoming calling. Recently we have experienced issues with certain calls not forwarding properly to mobile or remote employees as configured. An audit of our dial plan and call routing scripts is required to diagnose any flaws in how calls are being handled and directed. The successful candidate will have extensive familiarity with Asterisk and deep understanding of how call routing policies and SIP signaling interact. Temporary testing SIP accounts may need to be utilized to simulate various call scenarios and validate configurations are functioning as intended. Once areas of conflict or error are identified, modifications to our dial plan syntax, call handling scripts or related configuration files must be implemented. Thorough documentation of all analyses, findings and solution implementation is expected. Comprehensive testing will then be performed across our SIP trunks and phones to ensure call forwarding to alternate contacts functions seamlessly in all intended situations. Preference will be given to candidates holding the ATA Certification and with proven experience remediating complex VoIP setups involving multiple SIP providers and technologies like call queuing, logging and interactive voice response systems. Strong troubleshooting skills and methodical work approach is essential.
100.0 USD 技术与编程 peopleperhour 海外
1天前