SIP Trunk Requirements: Ports, Protocols, and Codecs
This article outlines the essential requirements for SIP trunks
Network Ports:
SIP Signaling (Port 5060): This port is crucial for establishing, managing, and terminating SIP calls. All SIP communication between your PBX and the Hosted Network SIP trunk uses UDP (User Datagram Protocol) on port 5060.
RTP Media (Ports 10000-20000): Real-time Transport Protocol (RTP) carries the actual voice data during a call. SIP trunking typically utilizes a range of UDP ports between 10000 and 20000 for sending and receiving RTP media streams.
Network Protocols:
UDP (User Datagram Protocol): SIP trunking relies on UDP for both SIP signaling (port 5060) and RTP media streams (ports 10000-20000)
Supported Codecs:
Codecs (coder/decoder) compress and decompress voice data for efficient transmission over the internet. Our SIP trunk supports various codecs,
G.711
G.729
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