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fix: Clarify SIP signaling IPs vs RTP media IPs in documentation #894

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fix: Clarify SIP signaling IPs vs RTP media IPs in documentation #894

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14 changes: 10 additions & 4 deletions fern/advanced/sip/sip-trunk.mdx
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Original file line number Diff line number Diff line change
Expand Up @@ -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.
Expand Down Expand Up @@ -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>


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