From e509d079a216fe92bd51846d839b492fb7f72d10 Mon Sep 17 00:00:00 2001
From: Winlin <winlinvip@gmail.com>
Date: Sun, 9 Nov 2025 12:11:50 -0500
Subject: [PATCH] Add codecs supprted in README

---
 README.md                             |  3 ++-
 trunk/3rdparty/srs-docs/doc/webrtc.md | 21 +++++++++++++++++++++
 2 files changed, 23 insertions(+), 1 deletion(-)

diff --git a/README.md b/README.md
index 46198b355..422a72cb3 100755
--- a/README.md
+++ b/README.md
@@ -11,7 +11,8 @@
 [![](https://codecov.io/gh/ossrs/srs/graph/badge.svg?token=Zx2LhdtA39)](https://codecov.io/gh/ossrs/srs)
 
 SRS/7.0 ([Kai](https://ossrs.io/lts/en-us/product#release-70)) is a simple, high-efficiency, and real-time video server, 
-supporting RTMP/WebRTC/HLS/HTTP-FLV/SRT/MPEG-DASH/GB28181, Linux/macOS, X86_64/ARMv7/AARCH64/M1/RISCV/LOONGARCH/MIPS, 
+supporting RTMP/WebRTC/HLS/HTTP-FLV/SRT/MPEG-DASH/GB28181, Linux/macOS, X86_64/ARMv7/AARCH64/M1/RISCV/LOONGARCH/MIPS,
+with codec support for H.264, H.265, AV1, VP9, AAC, Opus, and G.711, 
 and essential [features](trunk/doc/Features.md#features).
 
 [![SRS Overview](https://ossrs.net/wiki/images/SRS-SingleNode-4.0-sd.png?v=114)](https://ossrs.net/wiki/images/SRS-SingleNode-4.0-hd.png)
diff --git a/trunk/3rdparty/srs-docs/doc/webrtc.md b/trunk/3rdparty/srs-docs/doc/webrtc.md
index c72f40bd1..8f17c33c8 100644
--- a/trunk/3rdparty/srs-docs/doc/webrtc.md
+++ b/trunk/3rdparty/srs-docs/doc/webrtc.md
@@ -470,6 +470,27 @@ To use VP9, add the `codec=vp9` query parameter to WHIP/WHEP URLs:
 Browser support: Chrome/Edge M29+, Firefox M28+, Opera M16+. Safari does not support VP9. Use H.264 if you need RTMP/HLS conversion,
 DVR recording, or Safari compatibility.
 
+## G.711 Codec Support
+
+SRS supports G.711 (PCMU/PCMA) audio codec for WebRTC since v7.0.124 ([#4075](https://github.com/ossrs/srs/issues/4075)).
+G.711 is a widely-used, royalty-free audio codec with excellent compatibility across VoIP systems, IP cameras, and legacy telephony equipment.
+SRS implements G.711 as relay-only (SFU mode), accepting G.711 audio streams via WHIP and forwarding to WHEP players without transcoding.
+G.711 streams cannot be converted to RTMP/HLS or recorded to DVR.
+
+To use G.711, add the `acodec=pcmu` or `acodec=pcma` query parameter to WHIP/WHEP URLs:
+
+* Publish with PCMU: `http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream&acodec=pcmu`
+* Play with PCMU: `http://localhost:1985/rtc/v1/whep/?app=live&stream=livestream&acodec=pcmu`
+* Publish with PCMA: `http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream&acodec=pcma`
+
+You can also combine video and audio codec parameters:
+
+* VP9 + PCMU: `http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream&vcodec=vp9&acodec=pcmu`
+* H.264 + PCMA: `http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream&vcodec=h264&acodec=pcma`
+
+Browser support: All major browsers support G.711 (PCMU/PCMA). PCMU (μ-law) is prioritized over PCMA (A-law) when both are available.
+Use Opus if you need RTMP/HLS conversion or DVR recording.
+
 ## SFU: One to One
 
 Please use `conf/rtc.conf` as config.