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

Author: vapi-tasker[bot]

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
+
+<Warning>
+**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.
+</Warning>
 
 <Warning>
 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
     <Info>You might need to enable SIP REFER in your SIP provider to allow this.</Info>
   </Step>
 </Steps>
-
-