Ports and protocols
Well-known SIP ports come from the RFCs and IANA. Cloud ranges below were read from the vendor page on the date in this file. If a vendor edits the page, the page wins.
Rows
| Name | Port | Transport | Purpose | Note | Source |
|---|---|---|---|---|---|
| SIP | 5060 | UDP, TCP | Default port when a SIP URI has no port. | DNS SRV can publish a different port. RFC 3263. | Sources and go deeper
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| SIP over TLS | 5061 | TCP | Default port for a SIPS URI. | A trunk can listen elsewhere if DNS or the peer is explicit. | Sources and go deeper
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| STUN | 3478 | UDP, TCP | STUN binding and TURN allocation on the IANA stun port. | TURN is specified in RFC 8656 and uses this port unless configured otherwise. | Sources and go deeper
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| STUN/TURN over TLS | 5349 | TCP | IANA stuns / turns port. | Also registered for STUN over TLS. See RFC 8489. | Sources and go deeper
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| DNS | 53 | UDP, TCP | Name, NAPTR, and SRV lookups for SIP servers. | Large answers fall over to TCP. RFC 1035. | Sources and go deeper
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| RTP | negotiated | UDP | Media port is in SDP. There is no single IANA RTP port. | The usual convention is an even port. The SDP m= line is the authority. RFC 8866. | Sources and go deeper
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| RTCP | RTP + 1, or mux | UDP | Reports and feedback for the RTP stream. | Without mux or an a=rtcp line, RTCP is the next higher port. RFC 3550. | Sources and go deeper
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| MSRP | 2855 | TCP | IANA default for Message Session Relay Protocol. | Session setup is still SDP. The MSRP URI can carry another port. | Sources and go deeper
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| SIP over WebSocket | 80 or 443 | TCP | WebSocket is a transport for SIP, usually on HTTP(S) ports. | There is no dedicated IANA SIP-WebSocket port. | Sources and go deeper
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| NTP | 123 | UDP | Clock sync. TLS certificates and some phones depend on it. | Webex Calling also lists UDP 123 for devices. See the port reference. | Sources and go deeper
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| Teams Direct Routing signaling | 5061 | TCP (TLS) | SIP/TLS from the certified SBC to the Teams SIP proxy. | Read 2026-10-15. FQDNs: sip.pstnhub.microsoft.com, sip2, sip3. Return traffic uses the port configured on the SBC. Microsoft can change this page. | Sources and go deeper
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| Teams Direct Routing media | 3478-3481, 49152-53247 | UDP (SRTP) | Media between the SBC and Teams media processors. | Read 2026-10-15. Allow the published IP ranges, not only the addresses a single DNS answer returned. | Sources and go deeper
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| Webex Calling signaling (registration trunk and devices) | 8934 | TCP (TLS) | Outbound SIP-TLS to Webex Calling for registration-based trunks and devices. | Page change log dated 2026-09-07 expands phone source ports to 5060-5280. Local Gateway source is listed as 8000-65535. The help page wins if Cisco edits it. | Sources and go deeper
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| Webex Calling signaling (certificate trunk) | 5062 | TCP (TLS) | Outbound SIP-TLS for a certificate-based Local Gateway trunk. | Inbound from Webex to a certificate-based gateway uses a customer-chosen destination port. Registration-based gateways reuse the outbound connection. | Sources and go deeper
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| Webex Calling media | 5004, 9000, 8500-8699, 19560-65535 | UDP (SRTP) | Destination ports the port reference lists for SRTP toward Webex Calling. | Local Gateway source range on that page is UDP 8000-48199. Device source media is a narrower range (19560-19661 for many devices). Do not collapse those into one rule. | Sources and go deeper
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