RTP port ranges and media planning for VoIP

RTP carries the audio after SIP has negotiated a media address and port. RTP is usually accompanied by RTCP, and both are commonly carried over UDP. The exact range is implementation-specific and must be taken from the PBX, SBC, phone, or carrier configuration.

Why port ranges matter

A firewall rule that allows SIP on port 5060 does not automatically allow RTP. Media may use a large dynamic range, and NAT must keep the advertised and observed address mappings usable for the call duration.

Planning checklist

• Document each system’s configured RTP range.
• Confirm whether RTP and RTCP use adjacent ports or a different scheme.
• Estimate concurrent calls and sessions across every media edge.
• Allow only the required source and destination networks.
• Account for IPv4, IPv6, VPN, SBC, recording, and transcoding paths.
• Verify the range with a packet capture rather than relying on a vendor default.

One-way audio clue

Read the SDP address and port, then inspect the packet capture. If RTP leaves one endpoint but never arrives at the other, investigate routing, NAT, firewall policy, and address rewriting before changing codecs.

Related: VoIP protocols, NAT troubleshooting, and QoS.