SIP uses TCP and UDP protocols to carry its call control information (not the payload) and is usually carried on ports 5060 and 5061. The actual payload is transmitted using the RTP protocol (Real-time Transport Protocol) which is specifically designed to carry payloads that are time-sensitive information such as voice and video.
RTP has a broad range of ports assigned 16384 - 32767. However different SIP vendors use different ports they may or may not fall within this range.
Here are the ports needed for SIP to work.
• Call control: Ports 5060 and 5061
• RTP audio: Ports 16384 - 32767