Zoom Phone BYOC (Bring Your Own Carrier) lets you connect your existing SIP trunk to Zoom Phone. The SIP requirements are strict — TLS only, E.164 number format, specific codec profile — and failures are often silent. Here is how to configure it and fix the most common issues.
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Zoom Phone BYOC (Bring Your Own Carrier) allows you to connect your existing SIP trunk provider to Zoom Phone instead of using Zoom's own PSTN connectivity. You keep your existing carrier relationships, phone numbers, and rates while using Zoom Phone as the UCaaS platform.
BYOC works via a Session Border Controller (SBC) that sits between Zoom Phone and your SIP trunk carrier. The SBC handles protocol translation, security enforcement, and media handling. Zoom has a list of certified SBC partners, but any SBC that meets the technical requirements can work.
Two BYOC models exist:
| Requirement | Specification |
|---|---|
| Transport | TLS only (port 5061) — plain UDP/TCP not accepted |
| Media encryption | SRTP required |
| Certificate | Public CA signed — self-signed not accepted |
| Codecs | G.711 (PCMU/PCMA), G.722, G.729, Opus |
| Number format | E.164 (+12025551234) for all calls |
| DTMF | RFC 2833 (payload type 101) |
| OPTIONS keepalive | SBC must respond to Zoom OPTIONS pings |
| IP allowlisting | Allow Zoom SIP proxy IPs (published in Zoom docs) |
The SBC must handle two legs: the Zoom leg (TLS/SRTP, E.164) and the carrier leg (typically UDP, may be plain RTP, various number formats).
Zoom Phone requires E.164 format (+12025551234) for all calls. Most SIP carriers use 10-digit (2025551234) or 11-digit (12025551234) format. The SBC must normalize in both directions:
Zoom Admin Portal → Phone → SIP Trunk → select trunk → Status tab. Shows registration status, last OPTIONS ping time, and recent call records with SIP response codes.
In the capture, look for: TLS handshake completing (Client Hello → Server Hello → Finished), SIP INVITE messages with E.164 Request-URI, SDP with SRTP a=crypto lines on the Zoom leg, and RTCP RR (Receiver Reports) showing no packet loss for good calls.
Zoom Phone BYOC (Bring Your Own Carrier) lets you connect your existing SIP trunk provider to Zoom Phone instead of using Zoom's PSTN service. You keep your carrier, numbers, and rates while using Zoom Phone for the UCaaS platform. An SBC sits between Zoom and your carrier, handling TLS/SRTP on the Zoom side and your carrier's protocol on the other.
Zoom Phone BYOC requires TLS transport on port 5061 (plain UDP/TCP not accepted), SRTP for media, a public CA-signed certificate on your SBC, E.164 number format (+12025551234) for all calls, RFC 2833 DTMF, and the SBC must respond to OPTIONS keepalive pings from Zoom. Your firewall must allow inbound TLS 5061 from Zoom SIP proxy IP ranges.
Common Zoom BYOC call failures: SBC not reachable from Zoom (firewall blocking TLS 5061 or certificate invalid), codec mismatch where Zoom offers Opus but carrier only accepts G.711, number format mismatch where DID is not in E.164 format, or SRTP not configured on the carrier leg. Check the Zoom Admin Portal SIP Trunk status and capture SBC traces to identify the exact failure point.
Capture SIP from your SBC and upload to SIPSymposium. The analyzer checks TLS failures, codec negotiation, E.164 number format, SRTP configuration, and call flow between Zoom and your carrier.