---
id: "CVE-2026-116784"
kind: "vulnerability"
aliases: ["CVE-2026-116784"]
title: "FreeSWITCH: crafted RTCP packet can crash server and drop all calls"
severity: "high"
cvss: 7.5
epss: null
exploited: false
status: "open"
category: "runtime"
products: []
ecosystems: []
tags: ["freeswitch", "runtime", "telecom", "video-calls", "voip"]
not_affected_if: ["audio-only-calls", "version-1.11.2-or-later"]
published_at: "2026-11-05T20:17:45+00:00"
first_seen_at: "2026-11-05T20:40:45+00:00"
updated_at: "2026-11-05T20:43:04+00:00"
sources: [{"source": "nvd", "url": "https://nvd.nist.gov/vuln/detail/CVE-2026-116784", "first_seen_at": "2026-11-05T20:40:45+00:00"}]
exploit_sources: []
---

# FreeSWITCH: crafted RTCP packet can crash server and drop all calls

FreeSWITCH before 1.11.2 trusts an RTCP NACK packet's declared length when parsing feedback entries, so a short packet can make it read far past its 16 KiB receive buffer. A remote peer sending RTCP to an active video call can crash FreeSWITCH and end every call on it, though the flaw gives no write access.

_Summary, check and fix written by a model from the sources; labelled as such._

## How to check

- Check the FreeSWITCH version with `fs_cli -x version` or `freeswitch -version`; anything below `1.11.2` is affected.
- Check whether video is enabled for calls by looking for video codecs in your SIP profiles and dialplan.
- Note that calls can be affected even without an explicit RTCP interval set in the SIP profile, when remote SDP enables video RTCP.

## Fix

- FreeSWITCH: upgrade to `1.11.2`.

## Timeline

- 2026-11-05T20:40:45+00:00 nvd https://nvd.nist.gov/vuln/detail/CVE-2026-116784

## Documents

- [advisory] Vendor advisory: github.com https://github.com/signalwire/freeswitch/security/advisories/GHSA-f3xg-63wg-hjw4
- [fix] Patch: github.com https://github.com/signalwire/freeswitch/commit/e730a4e541b2cb49d54018a15f775728e19ce17f
- [fix] Release notes: github.com https://github.com/signalwire/freeswitch/releases/tag/v1.11.2

## From the sources

FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a software implementation that runs on any commodity hardware. Prior to 1.11.2, process_rtcp_report() used an RTPFB NACK packet's declared RTCP header length as its feedback-entry loop bound instead of the datagram length already validated by the caller. A short packet with an oversized declaration could drive reads far beyond the fixed 16 KiB receive buffer. Any peer able to send RTCP to an active video call can reach the flaw, including calls where remote SDP enables video RTCP even though the SIP profile has no explicit RTCP interval configured. The out-of-bounds read can crash FreeSWITCH and terminate every active call, but supplies no write primitive and does not echo the read values to the sender. This issue is fixed in version 1.11.2.

---

_Free to read, with fair use of about 60 pages a minute. For higher limits or API access: sales@infrastack.ai_