fix(webrtc): fix GStreamer WHEP interop (#4720)

## Summary

This PR fixes GStreamer interoperability issues during WebRTC/WHEP
negotiation with ZLMediaServer.

GStreamer could fail to establish the connection for two separate
reasons:

1. ZLMediaServer generated a non-compliant ICE `ufrag`. The generated
value contained `_`, which is not a valid ICE character, so GStreamer
rejected the SDP.
2. ZLMediaServer did not correctly handle `bundle-only` offers and could
answer an accepted bundled m-line with `port=0`, which caused the later
WHEP negotiation to fail.

## Changes

- Generate ICE `ufrag` values using ICE-compliant characters only.
- Preserve and handle `a=bundle-only` correctly during SDP parsing and
answer generation.
- Return `port=9` instead of `port=0` for accepted bundled m-lines.
- Add regression coverage for `bundle-only` SDP handling.
- URL-encode `delete_webrtc` query parameters in the returned `Location`
header so ICE-safe identifiers remain round-trippable over HTTP.

## Validation

- Built with WebRTC and SCTP enabled.
- Added regression test: `test_webrtc_regression`
- Verified:
  - ICE-safe identifier round-trip through `delete_webrtc`
  - `bundle-only` SDP answer generation

server/WebApi.cpp

@@ -2180,10 +2180,14 @@ void installWebApi() {
         WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, offer, headerOut, location](const WebRtcInterface &exchanger) mutable {
             auto &handler = const_cast<WebRtcInterface &>(exchanger);
             try {
+                // Encode query params since transport id/token may contain '+' or '/'.
+                HttpArgs delete_args;
+                delete_args["id"] = exchanger.getIdentifier();
+                delete_args["token"] = exchanger.deleteRandStr();
                 // 设置返回类型  [AUTO-TRANSLATED:ffc2a31a]
                 // Set return type
                 headerOut["Content-Type"] = "application/sdp";
-                headerOut["Location"] = location + "?id=" + exchanger.getIdentifier() + "&token=" + exchanger.deleteRandStr();
+                headerOut["Location"] = location + "?" + delete_args.make();
                 invoker(201, headerOut, handler.getAnswerSdp(offer));
             } catch (std::exception &ex) {
                 headerOut["Content-Type"] = "text/plain";

tests/CMakeLists.txt

@@ -37,7 +37,7 @@ foreach(TEST_SRC ${TEST_SRC_LIST})
 
   if(NOT TARGET ZLMediaKit::WebRTC)
     # 暂时过滤掉依赖 WebRTC 的测试模块
-    if("${TEST_EXE_NAME}" MATCHES "test_rtcp_nack")
+    if("${TEST_EXE_NAME}" MATCHES "test_rtcp_nack|test_webrtc_regression")
       continue()
     endif()
   endif()

tests/test_webrtc_regression.cpp

@@ -0,0 +1,115 @@
+/*
+ * Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
+ *
+ * Use of this source code is governed by MIT-like license that can be found in the
+ * LICENSE file in the root of the source tree. All contributing project authors
+ * may be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdlib>
+#include <iostream>
+#include <stdexcept>
+#include <string>
+
+#include "Common/Parser.h"
+#include "Common/strCoding.h"
+#include "Http/HttpClient.h"
+#include "../webrtc/Sdp.h"
+
+using namespace std;
+using namespace mediakit;
+
+namespace {
+
+void expect(bool cond, const string &msg) {
+    if (!cond) {
+        throw runtime_error(msg);
+    }
+}
+
+string makeBundleOnlyDatachannelOffer() {
+    return
+        "v=0\r\n"
+        "o=- 0 0 IN IP4 127.0.0.1\r\n"
+        "s=-\r\n"
+        "t=0 0\r\n"
+        "a=group:BUNDLE 0\r\n"
+        "a=msid-semantic: WMS\r\n"
+        "m=application 0 UDP/DTLS/SCTP webrtc-datachannel\r\n"
+        "c=IN IP4 0.0.0.0\r\n"
+        "a=ice-ufrag:remoteUfrag\r\n"
+        "a=ice-pwd:remotePassword1234567890\r\n"
+        "a=fingerprint:sha-256 "
+        "00:11:22:33:44:55:66:77:88:99:AA:BB:CC:DD:EE:FF:"
+        "00:11:22:33:44:55:66:77:88:99:AA:BB:CC:DD:EE:FF\r\n"
+        "a=setup:actpass\r\n"
+        "a=mid:0\r\n"
+        "a=bundle-only\r\n"
+        "a=sctp-port:5000\r\n";
+}
+
+void testBundleOnlyDatachannelAnswer() {
+    RtcSession offer;
+    offer.loadFrom(makeBundleOnlyDatachannelOffer());
+    offer.checkValid();
+
+    expect(offer.media.size() == 1, "offer should contain a single application m-line");
+    expect(offer.media[0].bundle_only, "offer application m-line should preserve a=bundle-only");
+
+    SdpAttrFingerprint local_fingerprint;
+    local_fingerprint.algorithm = "sha-256";
+    local_fingerprint.hash =
+        "FF:EE:DD:CC:BB:AA:99:88:77:66:55:44:33:22:11:00:"
+        "FF:EE:DD:CC:BB:AA:99:88:77:66:55:44:33:22:11:00";
+
+    RtcConfigure configure;
+    configure.setDefaultSetting("localUfrag", "localPassword1234567890", RtpDirection::sendrecv, local_fingerprint);
+
+    auto answer = configure.createAnswer(offer);
+    expect(answer != nullptr, "createAnswer should return a session");
+    expect(answer->media.size() == 1, "answer should contain a single application m-line");
+
+#ifdef ENABLE_SCTP
+    answer->checkValid();
+    expect(answer->media[0].port == 9, "bundle-only application m-line should use port 9 in answer");
+    expect(answer->group.mids.size() == 1 && answer->group.mids[0] == "0",
+           "accepted bundle-only application m-line should remain in group:BUNDLE");
+#else
+    expect(answer->media[0].port == 0, "application m-line should stay rejected when SCTP is disabled");
+#endif
+}
+
+void testDeleteWebrtcLocationQueryRoundTrip() {
+    const string raw_id = "Ab+/9";
+    const string raw_token = "token+/9";
+
+    HttpArgs args;
+    args["id"] = raw_id;
+    args["token"] = raw_token;
+    auto query = args.make();
+
+    expect(query.find("id=Ab%2B%2F9") != string::npos, "id should be URL-encoded in delete_webrtc query");
+    expect(query.find("token=token%2B%2F9") != string::npos,
+           "token should be URL-encoded in delete_webrtc query");
+
+    auto parsed = Parser::parseArgs(query);
+    expect(strCoding::UrlDecodeComponent(parsed["id"]) == raw_id, "encoded id should round-trip through query parsing");
+    expect(strCoding::UrlDecodeComponent(parsed["token"]) == raw_token,
+           "encoded token should round-trip through query parsing");
+}
+
+} // namespace
+
+int main() {
+    try {
+        testBundleOnlyDatachannelAnswer();
+        testDeleteWebrtcLocationQueryRoundTrip();
+        cout << "test_webrtc_regression passed" << endl;
+        return 0;
+    } catch (const exception &ex) {
+        cerr << "test_webrtc_regression failed: " << ex.what() << endl;
+        return EXIT_FAILURE;
+    }
+}

webrtc/Sdp.cpp

@@ -799,6 +799,7 @@ void RtcSession::loadFrom(const string &str) {
             rtc_media.fingerprint = sdp.getItemClass<SdpAttrFingerprint>('a', "fingerprint");
         }
         rtc_media.ice_lite = media.getItem('a', "ice-lite").operator bool();
+        rtc_media.bundle_only = media.getItem('a', "bundle-only").operator bool();
         auto ice_options = media.getItemClass<SdpAttrIceOption>('a', "ice-options");
         rtc_media.ice_trickle = ice_options.trickle;
         rtc_media.ice_renomination = ice_options.renomination;
@@ -1659,6 +1660,15 @@ static DtlsRole mathDtlsRole(DtlsRole role) {
     }
 }
 
+static uint16_t getAnswerMediaPort(const RtcMedia &offer_media) {
+    // RFC 8843: bundle-only m-lines may use port=0 in the offer but still need a
+    // real port placeholder in the accepted answer.
+    if (!offer_media.port && offer_media.bundle_only) {
+        return 9;
+    }
+    return offer_media.port;
+}
+
 void RtcConfigure::createMediaOffer(const std::shared_ptr<RtcSession> &ret) const {
     int index = 0;
     if (video.direction != RtpDirection::sendonly || _rtsp_video_plan)  {
@@ -1826,6 +1836,7 @@ RETRY:
 
     if (offer_media.type == TrackApplication) {
         RtcMedia answer_media = offer_media;
+        answer_media.port = getAnswerMediaPort(offer_media);
         answer_media.role = mathDtlsRole(offer_media.role);
         answer_media.ice_ufrag = configure.ice_ufrag;
         answer_media.ice_pwd = configure.ice_pwd;
@@ -1871,7 +1882,7 @@ RETRY:
         answer_media.type = offer_media.type;
         answer_media.mid = offer_media.mid;
         answer_media.proto = offer_media.proto;
-        answer_media.port = offer_media.port;
+        answer_media.port = getAnswerMediaPort(offer_media);
         answer_media.addr = offer_media.addr;
         answer_media.bandwidth = offer_media.bandwidth;
         answer_media.rtcp_addr = offer_media.rtcp_addr;

webrtc/Sdp.h

@@ -652,6 +652,9 @@ public:
     bool rtcp_rsize { false };
     SdpAttrRtcp rtcp_addr;
 
+    //////// bundle ////////
+    bool bundle_only { false };
+
     //////// ice ////////
     bool ice_trickle { false };
     bool ice_lite { false };

webrtc/WebRtcTransport.cpp

@@ -137,7 +137,12 @@ static std::string getServerPrefix() {
     // 拷贝tcp端口  [AUTO-TRANSLATED:23191878]
     // Copy tcp port
     memcpy(buf + 6, &(reinterpret_cast<sockaddr_in *>(&addr)->sin_port), 2);
-    auto ret = encodeBase64(string(buf, 8)) + '_';
+    // RFC 5245 §15.4: ice-char = ALPHA / DIGIT / "+" / "/"
+    auto ret = encodeBase64(string(buf, 8));
+    // Remove base64 '=' padding (not a valid ice-char)
+    ret.erase(std::remove(ret.begin(), ret.end(), '='), ret.end());
+    // Use '/' separator instead of '_' (not a valid ice-char)
+    ret += '/';
     InfoL << "MediaServer(" << host << ":" << udp_port << ":" << tcp_port << ") prefix: " << ret;
     return ret;
 }