From c5cbc2a8ce729f8b3a5279bbfbe2f214bb2afd9f Mon Sep 17 00:00:00 2001 From: "vapi-tasker[bot]" <253425205+vapi-tasker[bot]@users.noreply.github.com> Date: Mon, 26 Jan 2026 07:03:41 +0000 Subject: [PATCH] fix: clarify SIP signaling IPs vs RTP media IPs in documentation The documentation previously implied that the listed IP addresses (44.229.228.186/32, 44.238.177.138/32) were for both SIP signaling and RTP media traffic. This update clarifies that: 1. These IPs are for SIP signaling ONLY 2. RTP media traffic does not currently have a static set of IPs that can be whitelisted This helps customers understand that they cannot whitelist specific IPs for RTP media traffic at this time. Resolves VAP-11461 --- fern/advanced/sip/sip-trunk.mdx | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/fern/advanced/sip/sip-trunk.mdx b/fern/advanced/sip/sip-trunk.mdx index 0e40db6fb..bebb3367f 100644 --- a/fern/advanced/sip/sip-trunk.mdx +++ b/fern/advanced/sip/sip-trunk.mdx @@ -8,12 +8,20 @@ SIP trunking replaces traditional phone lines with a virtual connection over the ## Network requirements -To allow SIP signaling and media between Vapi and your SIP provider, you must allowlist the following IP addresses: +### SIP signaling + +To allow SIP signaling between Vapi and your SIP provider, you must allowlist the following IP addresses: - 44.229.228.186/32 - 44.238.177.138/32 -These IPs are used exclusively for SIP traffic. +These IPs are used exclusively for **SIP signaling traffic only** (call setup, teardown, and control messages). + +### RTP media + + +**RTP media traffic does not currently have a static set of IP addresses that can be whitelisted.** RTP media (the actual audio/video streams) may originate from dynamic IP addresses. If your firewall requires specific IP allowlisting for RTP traffic, please contact Vapi support to discuss your requirements. + We generally don't recommend IP-based authentication for SIP trunks as it can lead to routing issues. Since our servers are shared by many customers, if your telephony provider has multiple customers using IP-based authentication, calls may be routed incorrectly. IP-based authentication works reliably only when your SIP provider offers a unique termination URI or a dedicated SIP server for each customer, as is the case with Plivo and Twilio integrations. @@ -130,5 +138,3 @@ Note: Certain providers require phone numbers to be formatted in the proper E.16 You might need to enable SIP REFER in your SIP provider to allow this. - -