diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
index 1082dad67..1f9da26b2 100644
--- a/trunk/conf/full.conf
+++ b/trunk/conf/full.conf
@@ -434,6 +434,10 @@ rtc_server {
     #       *           Retrieve server IP automatically, from all network interfaces.
     #       $CANDIDATE  Read the IP from ENV variable, use * if not set.
     #       x.x.x.x     A specified IP address or DNS name, use * if 0.0.0.0.
+    # You can also set the candidate by the query string eip, note that you can also set the UDP port,
+    # for example:
+    #       http://locahost:1985/rtc/v1/whip/?app=live&stream=livestream&eip=192.168.3.11
+    #       http://locahost:1985/rtc/v1/whip/?app=live&stream=livestream&eip=192.168.3.11:18000
     # @remark For Firefox, the candidate MUST be IP, MUST NOT be DNS name, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
     # @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
     # Overwrite by env SRS_RTC_SERVER_CANDIDATE
diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md
index 28b8805b6..34e358167 100644
--- a/trunk/doc/CHANGELOG.md
+++ b/trunk/doc/CHANGELOG.md
@@ -7,6 +7,7 @@ The changelog for SRS.
 <a name="v6-changes"></a>
 
 ## SRS 6.0 Changelog
+* v6.0, 2024-02-06, Merge [#3920](https://github.com/ossrs/srs/pull/3920): WHIP: Fix bug for converting WHIP to RTMP/HLS. v6.0.113 (#3920)
 * v6.0, 2024-02-05, Merge [#3924](https://github.com/ossrs/srs/pull/3924): Upgrade hls.js and set in low latency mode. v6.0.112 (#3924)
 * v6.0, 2024-02-05, Merge [#3925](https://github.com/ossrs/srs/pull/3925): RTC: Fix video and audio track pt_ is not change in player before publisher. v6.0.111 (#3925)
 * v6.0, 2024-02-05, Merge [#3923](https://github.com/ossrs/srs/pull/3923): Configure: print enabled/disable sanitizer. v6.0.110 (#3923)
@@ -123,6 +124,7 @@ The changelog for SRS.
 <a name="v5-changes"></a>
 
 ## SRS 5.0 Changelog
+* v5.0, 2024-02-06, Merge [#3920](https://github.com/ossrs/srs/pull/3920): WHIP: Fix bug for converting WHIP to RTMP/HLS. v5.0.208 (#3920)
 * v5.0, 2024-02-05, Merge [#3925](https://github.com/ossrs/srs/pull/3925): RTC: Fix video and audio track pt_ is not change in player before publisher. v5.0.207 (#3925)
 * v5.0, 2024-02-05, Merge [#3923](https://github.com/ossrs/srs/pull/3923): Configure: print enabled/disable sanitizer. v5.0.206 (#3923)
 * v5.0, 2023-12-30, Merge [#3916](https://github.com/ossrs/srs/pull/3916): Enhancing the compatibility of options.sh. v5.0.204 (#3916)
diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp
index 89377e728..c74b59e9e 100644
--- a/trunk/src/app/srs_app_rtc_source.cpp
+++ b/trunk/src/app/srs_app_rtc_source.cpp
@@ -1318,12 +1318,16 @@ SrsRtcFrameBuilder::SrsRtcFrameBuilder(ISrsStreamBridge* bridge)
     header_sn_ = 0;
     memset(cache_video_pkts_, 0, sizeof(cache_video_pkts_));
     rtp_key_frame_ts_ = -1;
+    sync_state_ = -1;
+    obs_whip_sps_ = obs_whip_pps_ = NULL;
 }
 
 SrsRtcFrameBuilder::~SrsRtcFrameBuilder()
 {
     srs_freep(codec_);
     clear_cached_video();
+    srs_freep(obs_whip_sps_);
+    srs_freep(obs_whip_pps_);
 }
 
 srs_error_t SrsRtcFrameBuilder::initialize(SrsRequest* r)
@@ -1366,8 +1370,18 @@ srs_error_t SrsRtcFrameBuilder::on_rtp(SrsRtpPacket *pkt)
 
     // Have no received any sender report, can't calculate avsync_time,
     // discard it to avoid timestamp problem in live source
+    const SrsRtpHeader& h = pkt->header;
     if (pkt->get_avsync_time() <= 0) {
+        if (sync_state_ < 0) {
+            srs_trace("RTC: Discard no-sync %s, ssrc=%u, seq=%u, ts=%u, state=%d", pkt->is_audio() ? "Audio" : "Video",
+                h.get_ssrc(), h.get_sequence(), h.get_timestamp(), sync_state_);
+            sync_state_ = 0;
+        }
         return err;
+    } else if (sync_state_ < 1) {
+        srs_trace("RTC: Accept sync %s, ssrc=%u, seq=%u, ts=%u, state=%d", pkt->is_audio() ? "Audio" : "Video",
+            h.get_ssrc(), h.get_sequence(), h.get_timestamp(), sync_state_);
+        sync_state_ = 2;
     }
 
     if (pkt->is_audio()) {
@@ -1499,46 +1513,75 @@ srs_error_t SrsRtcFrameBuilder::packet_video_key_frame(SrsRtpPacket* pkt)
 {
     srs_error_t err = srs_success;
 
-    // TODO: handle sps and pps in 2 rtp packets
+    // For OBS WHIP, it uses RTP Raw packet with SPS/PPS/IDR frame. Note that not all
+    // raw payload is SPS/PPS.
+    bool has_sps_pps_in_raw_payload = false;
+    SrsRtpRawPayload* raw_payload = dynamic_cast<SrsRtpRawPayload*>(pkt->payload());
+    if (raw_payload) {
+        if (pkt->nalu_type == SrsAvcNaluTypeSPS) {
+            has_sps_pps_in_raw_payload = true;
+            srs_freep(obs_whip_sps_);
+            obs_whip_sps_ = pkt->copy();
+        } else if (pkt->nalu_type == SrsAvcNaluTypePPS) {
+            has_sps_pps_in_raw_payload = true;
+            srs_freep(obs_whip_pps_);
+            obs_whip_pps_ = pkt->copy();
+        }
+        // Ignore if one of OBS WHIP SPS/PPS is not ready.
+        if (has_sps_pps_in_raw_payload && (!obs_whip_sps_ || !obs_whip_pps_)) {
+            return err;
+        }
+    }
+
+    // Generally, there will be SPS+PPS+IDR in a STAP-A packet.
     SrsRtpSTAPPayload* stap_payload = dynamic_cast<SrsRtpSTAPPayload*>(pkt->payload());
-    if (stap_payload) {
-        SrsSample* sps = stap_payload->get_sps();
-        SrsSample* pps = stap_payload->get_pps();
-        if (NULL == sps || NULL == pps) {
+
+    // Handle SPS/PPS in cache or STAP-A packet.
+    if (stap_payload || has_sps_pps_in_raw_payload) {
+        // Get the SPS/PPS from cache or STAP-A packet.
+        SrsSample* sps = stap_payload ? stap_payload->get_sps() : NULL;
+        if (!sps && obs_whip_sps_) sps = dynamic_cast<SrsRtpRawPayload*>(obs_whip_sps_->payload())->sample_;
+        SrsSample* pps = stap_payload ? stap_payload->get_pps() : NULL;
+        if (!pps && obs_whip_pps_) pps = dynamic_cast<SrsRtpRawPayload*>(obs_whip_pps_->payload())->sample_;
+        if (!sps || !pps) {
             return srs_error_new(ERROR_RTC_RTP_MUXER, "no sps or pps in stap-a rtp. sps: %p, pps:%p", sps, pps);
-        } else {
-            // h264 raw to h264 packet.
-            std::string sh;
-            SrsRawH264Stream* avc = new SrsRawH264Stream();
-            SrsAutoFree(SrsRawH264Stream, avc);
+        }
 
-            if ((err = avc->mux_sequence_header(string(sps->bytes, sps->size), string(pps->bytes, pps->size), sh)) != srs_success) {
-                return srs_error_wrap(err, "mux sequence header");
-            }
+        // Reset SPS/PPS cache, ensuring that the next SPS/PPS will be handled when both are received.
+        SrsAutoFree(SrsRtpPacket, obs_whip_sps_);
+        SrsAutoFree(SrsRtpPacket, obs_whip_pps_);
 
-            // h264 packet to flv packet.
-            char* flv = NULL;
-            int nb_flv = 0;
-            if ((err = avc->mux_avc2flv(sh, SrsVideoAvcFrameTypeKeyFrame, SrsVideoAvcFrameTraitSequenceHeader, pkt->get_avsync_time(),
-                                        pkt->get_avsync_time(), &flv, &nb_flv)) != srs_success) {
-                return srs_error_wrap(err, "avc to flv");
-            }
+        // h264 raw to h264 packet.
+        std::string sh;
+        SrsRawH264Stream* avc = new SrsRawH264Stream();
+        SrsAutoFree(SrsRawH264Stream, avc);
 
-            SrsMessageHeader header;
-            header.initialize_video(nb_flv, pkt->get_avsync_time(), 1);
-            SrsCommonMessage rtmp;
-            if ((err = rtmp.create(&header, flv, nb_flv)) != srs_success) {
-                return srs_error_wrap(err, "create rtmp");
-            }
+        if ((err = avc->mux_sequence_header(string(sps->bytes, sps->size), string(pps->bytes, pps->size), sh)) != srs_success) {
+            return srs_error_wrap(err, "mux sequence header");
+        }
 
-            SrsSharedPtrMessage msg;
-            if ((err = msg.create(&rtmp)) != srs_success) {
-                return srs_error_wrap(err, "create message");
-            }
+        // h264 packet to flv packet.
+        char* flv = NULL;
+        int nb_flv = 0;
+        if ((err = avc->mux_avc2flv(sh, SrsVideoAvcFrameTypeKeyFrame, SrsVideoAvcFrameTraitSequenceHeader, pkt->get_avsync_time(),
+                                    pkt->get_avsync_time(), &flv, &nb_flv)) != srs_success) {
+            return srs_error_wrap(err, "avc to flv");
+        }
 
-            if ((err = bridge_->on_frame(&msg)) != srs_success) {
-                return err;
-            }
+        SrsMessageHeader header;
+        header.initialize_video(nb_flv, pkt->get_avsync_time(), 1);
+        SrsCommonMessage rtmp;
+        if ((err = rtmp.create(&header, flv, nb_flv)) != srs_success) {
+            return srs_error_wrap(err, "create rtmp");
+        }
+
+        SrsSharedPtrMessage msg;
+        if ((err = msg.create(&rtmp)) != srs_success) {
+            return srs_error_wrap(err, "create message");
+        }
+
+        if ((err = bridge_->on_frame(&msg)) != srs_success) {
+            return err;
         }
     }
 
@@ -1583,7 +1626,7 @@ srs_error_t SrsRtcFrameBuilder::packet_video_key_frame(SrsRtpPacket* pkt)
     if (-1 == sn) {
         if (check_frame_complete(header_sn_, tail_sn)) {
             if ((err = packet_video_rtmp(header_sn_, tail_sn)) != srs_success) {
-                err = srs_error_wrap(err, "fail to packet key frame");
+                err = srs_error_wrap(err, "fail to packet frame");
             }
         }
     } else if (-2 == sn) {
diff --git a/trunk/src/app/srs_app_rtc_source.hpp b/trunk/src/app/srs_app_rtc_source.hpp
index dca62a59e..02e786cf2 100644
--- a/trunk/src/app/srs_app_rtc_source.hpp
+++ b/trunk/src/app/srs_app_rtc_source.hpp
@@ -318,6 +318,13 @@ private:
     uint16_t header_sn_;
     uint16_t lost_sn_;
     int64_t rtp_key_frame_ts_;
+private:
+    // The state for timestamp sync state. -1 for init. 0 not sync. 1 sync.
+    int sync_state_;
+private:
+    // For OBS WHIP, send SPS/PPS in dedicated RTP packet.
+    SrsRtpPacket* obs_whip_sps_;
+    SrsRtpPacket* obs_whip_pps_;
 public:
     SrsRtcFrameBuilder(ISrsStreamBridge* bridge);
     virtual ~SrsRtcFrameBuilder();
diff --git a/trunk/src/core/srs_core_version5.hpp b/trunk/src/core/srs_core_version5.hpp
index 4fce369da..1e9a74079 100644
--- a/trunk/src/core/srs_core_version5.hpp
+++ b/trunk/src/core/srs_core_version5.hpp
@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       5
 #define VERSION_MINOR       0
-#define VERSION_REVISION    207
+#define VERSION_REVISION    208
 
 #endif
diff --git a/trunk/src/core/srs_core_version6.hpp b/trunk/src/core/srs_core_version6.hpp
index b4a595697..d19e4628a 100644
--- a/trunk/src/core/srs_core_version6.hpp
+++ b/trunk/src/core/srs_core_version6.hpp
@@ -9,6 +9,6 @@
 
 #define VERSION_MAJOR       6
 #define VERSION_MINOR       0
-#define VERSION_REVISION    112
+#define VERSION_REVISION    113
 
 #endif
diff --git a/trunk/src/kernel/srs_kernel_rtc_rtp.cpp b/trunk/src/kernel/srs_kernel_rtc_rtp.cpp
index 3dec5de0f..761223f8b 100644
--- a/trunk/src/kernel/srs_kernel_rtc_rtp.cpp
+++ b/trunk/src/kernel/srs_kernel_rtc_rtp.cpp
@@ -970,12 +970,14 @@ SrsRtpRawPayload::SrsRtpRawPayload()
 {
     payload = NULL;
     nn_payload = 0;
+    sample_ = new SrsSample();
 
     ++_srs_pps_objs_rraw->sugar;
 }
 
 SrsRtpRawPayload::~SrsRtpRawPayload()
 {
+    srs_freep(sample_);
 }
 
 uint64_t SrsRtpRawPayload::nb_bytes()
@@ -1007,6 +1009,9 @@ srs_error_t SrsRtpRawPayload::decode(SrsBuffer* buf)
     payload = buf->head();
     nn_payload = buf->left();
 
+    sample_->bytes = payload;
+    sample_->size = nn_payload;
+
     return srs_success;
 }
 
@@ -1016,6 +1021,7 @@ ISrsRtpPayloader* SrsRtpRawPayload::copy()
 
     cp->payload = payload;
     cp->nn_payload = nn_payload;
+    cp->sample_ = sample_->copy();
 
     return cp;
 }
diff --git a/trunk/src/kernel/srs_kernel_rtc_rtp.hpp b/trunk/src/kernel/srs_kernel_rtc_rtp.hpp
index 9621ac035..9aca86e66 100644
--- a/trunk/src/kernel/srs_kernel_rtc_rtp.hpp
+++ b/trunk/src/kernel/srs_kernel_rtc_rtp.hpp
@@ -348,6 +348,9 @@ public:
     // @remark We only refer to the memory, user must free its bytes.
     char* payload;
     int nn_payload;
+public:
+    // Use the whole RAW RTP payload as a sample.
+    SrsSample* sample_;
 public:
     SrsRtpRawPayload();
     virtual ~SrsRtpRawPayload();