From: baigao via ffmpeg-devel <ffmpeg-devel@ffmpeg.org>
To: ffmpeg-devel@ffmpeg.org
Cc: baigao <1007668733@qq.com>
Subject: [FFmpeg-devel] [PATCH 1/3] avformat/whip whep: create rtc for common RTC code shared by whip and whep
Date: Sun, 12 Oct 2025 23:41:49 +0800
Message-ID: <tencent_E2B214F178ED5D74001337140E6BC623D905@qq.com> (raw)
In-Reply-To: <20251012152347.1022477-1-1007668733@qq.com>
---
libavformat/Makefile | 2 +-
libavformat/{whip.c => rtc.c} | 856 +-------------------
libavformat/rtc.h | 220 ++++++
libavformat/whip.c | 1386 +--------------------------------
4 files changed, 264 insertions(+), 2200 deletions(-)
copy libavformat/{whip.c => rtc.c} (59%)
create mode 100644 libavformat/rtc.h
diff --git a/libavformat/Makefile b/libavformat/Makefile
index ed93458f03..9261245755 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -640,7 +640,7 @@ OBJS-$(CONFIG_WEBM_CHUNK_MUXER) += webm_chunk.o
OBJS-$(CONFIG_WEBP_MUXER) += webpenc.o
OBJS-$(CONFIG_WEBVTT_DEMUXER) += webvttdec.o subtitles.o
OBJS-$(CONFIG_WEBVTT_MUXER) += webvttenc.o
-OBJS-$(CONFIG_WHIP_MUXER) += whip.o avc.o http.o srtp.o
+OBJS-$(CONFIG_WHIP_MUXER) += whip.o rtc.o avc.o http.o srtp.o
OBJS-$(CONFIG_WSAUD_DEMUXER) += westwood_aud.o
OBJS-$(CONFIG_WSAUD_MUXER) += westwood_audenc.o
OBJS-$(CONFIG_WSD_DEMUXER) += wsddec.o rawdec.o
diff --git a/libavformat/whip.c b/libavformat/rtc.c
similarity index 59%
copy from libavformat/whip.c
copy to libavformat/rtc.c
index e809075643..2dc0383d3e 100644
--- a/libavformat/whip.c
+++ b/libavformat/rtc.c
@@ -1,5 +1,5 @@
/*
- * WebRTC-HTTP ingestion protocol (WHIP) muxer
+ * WebRTC protocol
* Copyright (c) 2023 The FFmpeg Project
*
* This file is part of FFmpeg.
@@ -19,30 +19,19 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#include "libavcodec/avcodec.h"
-#include "libavcodec/codec_desc.h"
-#include "libavcodec/h264.h"
-#include "libavcodec/startcode.h"
-#include "libavutil/base64.h"
-#include "libavutil/bprint.h"
+#include "libavutil/time.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/random_seed.h"
#include "libavutil/crc.h"
#include "libavutil/hmac.h"
-#include "libavutil/intreadwrite.h"
-#include "libavutil/lfg.h"
-#include "libavutil/opt.h"
#include "libavutil/mem.h"
-#include "libavutil/random_seed.h"
-#include "libavutil/time.h"
-#include "avc.h"
-#include "nal.h"
+#include "libavutil/base64.h"
+
#include "avio_internal.h"
-#include "http.h"
#include "internal.h"
-#include "mux.h"
#include "network.h"
-#include "rtp.h"
-#include "srtp.h"
-#include "tls.h"
+#include "http.h"
+#include "rtc.h"
/**
* Maximum size limit of a Session Description Protocol (SDP),
@@ -59,16 +48,6 @@
#define DTLS_SRTP_KEY_LEN 16
#define DTLS_SRTP_SALT_LEN 14
-/**
- * The maximum size of the Secure Real-time Transport Protocol (SRTP) HMAC checksum
- * and padding that is appended to the end of the packet. To calculate the maximum
- * size of the User Datagram Protocol (UDP) packet that can be sent out, subtract
- * this size from the `pkt_size`.
- */
-#define DTLS_SRTP_CHECKSUM_LEN 16
-
-#define WHIP_US_PER_MS 1000
-
/**
* If we try to read from UDP and get EAGAIN, we sleep for 5ms and retry up to 10 times.
* This will limit the total duration (in milliseconds, 50ms)
@@ -130,26 +109,6 @@
*/
#define ICE_STUN_HEADER_SIZE 20
-/**
- * The RTP header is 12 bytes long, comprising the Version(1B), PT(1B),
- * SequenceNumber(2B), Timestamp(4B), and SSRC(4B).
- * See https://www.rfc-editor.org/rfc/rfc3550#section-5.1
- */
-#define WHIP_RTP_HEADER_SIZE 12
-
-/**
- * For RTCP, PT is [128, 223] (or without marker [0, 95]). Literally, RTCP starts
- * from 64 not 0, so PT is [192, 223] (or without marker [64, 95]), see "RTCP Control
- * Packet Types (PT)" at
- * https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-4
- *
- * For RTP, the PT is [96, 127], or [224, 255] with marker. See "RTP Payload Types (PT)
- * for standard audio and video encodings" at
- * https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-1
- */
-#define WHIP_RTCP_PT_START 192
-#define WHIP_RTCP_PT_END 223
-
/**
* In the case of ICE-LITE, these fields are not used; instead, they are defined
* as constant values.
@@ -157,17 +116,6 @@
#define WHIP_SDP_SESSION_ID "4489045141692799359"
#define WHIP_SDP_CREATOR_IP "127.0.0.1"
-/**
- * Refer to RFC 7675 5.1,
- *
- * To prevent expiry of consent, a STUN binding request can be sent periodically.
- * Implementations SHOULD set a default interval of 5 seconds(5000ms).
- *
- * Consent expires after 30 seconds(30000ms).
- */
-#define WHIP_ICE_CONSENT_CHECK_INTERVAL 5000
-#define WHIP_ICE_CONSENT_EXPIRED_TIMER 30000
-
/* Calculate the elapsed time from starttime to endtime in milliseconds. */
#define ELAPSED(starttime, endtime) ((float)(endtime - starttime) / 1000)
@@ -181,167 +129,16 @@ enum STUNAttr {
STUN_ATTR_ICE_CONTROLLING = 0x802A, /// ICE controlling role
};
-enum WHIPState {
- WHIP_STATE_NONE,
-
- /* The initial state. */
- WHIP_STATE_INIT,
- /* The muxer has sent the offer to the peer. */
- WHIP_STATE_OFFER,
- /* The muxer has received the answer from the peer. */
- WHIP_STATE_ANSWER,
- /**
- * After parsing the answer received from the peer, the muxer negotiates the abilities
- * in the offer that it generated.
- */
- WHIP_STATE_NEGOTIATED,
- /* The muxer has connected to the peer via UDP. */
- WHIP_STATE_UDP_CONNECTED,
- /* The muxer has sent the ICE request to the peer. */
- WHIP_STATE_ICE_CONNECTING,
- /* The muxer has received the ICE response from the peer. */
- WHIP_STATE_ICE_CONNECTED,
- /* The muxer has finished the DTLS handshake with the peer. */
- WHIP_STATE_DTLS_FINISHED,
- /* The muxer has finished the SRTP setup. */
- WHIP_STATE_SRTP_FINISHED,
- /* The muxer is ready to send/receive media frames. */
- WHIP_STATE_READY,
- /* The muxer is failed. */
- WHIP_STATE_FAILED,
-};
-
-typedef struct WHIPContext {
- AVClass *av_class;
-
- /* The state of the RTC connection. */
- enum WHIPState state;
-
- /* Parameters for the input audio and video codecs. */
- AVCodecParameters *audio_par;
- AVCodecParameters *video_par;
-
- /**
- * The h264_mp4toannexb Bitstream Filter (BSF) bypasses the AnnexB packet;
- * therefore, it is essential to insert the SPS and PPS before each IDR frame
- * in such cases.
- */
- int h264_annexb_insert_sps_pps;
-
- /* The random number generator. */
- AVLFG rnd;
-
- /* The ICE username and pwd fragment generated by the muxer. */
- char ice_ufrag_local[9];
- char ice_pwd_local[33];
- /* The SSRC of the audio and video stream, generated by the muxer. */
- uint32_t audio_ssrc;
- uint32_t video_ssrc;
- uint32_t video_rtx_ssrc;
-
- uint16_t audio_first_seq;
- uint16_t video_first_seq;
- /* The PT(Payload Type) of stream, generated by the muxer. */
- uint8_t audio_payload_type;
- uint8_t video_payload_type;
- uint8_t video_rtx_payload_type;
- /**
- * This is the SDP offer generated by the muxer based on the codec parameters,
- * DTLS, and ICE information.
- */
- char *sdp_offer;
-
- int is_peer_ice_lite;
- uint64_t ice_tie_breaker; // random 64 bit, for ICE-CONTROLLING
- /* The ICE username and pwd from remote server. */
- char *ice_ufrag_remote;
- char *ice_pwd_remote;
- /**
- * This represents the ICE candidate protocol, priority, host and port.
- * Currently, we only support one candidate and choose the first UDP candidate.
- * However, we plan to support multiple candidates in the future.
- */
- char *ice_protocol;
- char *ice_host;
- int ice_port;
-
- /* The SDP answer received from the WebRTC server. */
- char *sdp_answer;
- /* The resource URL returned in the Location header of WHIP HTTP response. */
- char *whip_resource_url;
-
- /* These variables represent timestamps used for calculating and tracking the cost. */
- int64_t whip_starttime;
- int64_t whip_init_time;
- int64_t whip_offer_time;
- int64_t whip_answer_time;
- int64_t whip_udp_time;
- int64_t whip_ice_time;
- int64_t whip_dtls_time;
- int64_t whip_srtp_time;
- int64_t whip_last_consent_tx_time;
- int64_t whip_last_consent_rx_time;
-
- /* The certificate and private key content used for DTLS handshake */
- char cert_buf[MAX_CERTIFICATE_SIZE];
- char key_buf[MAX_CERTIFICATE_SIZE];
- /* The fingerprint of certificate, used in SDP offer. */
- char *dtls_fingerprint;
- /**
- * This represents the material used to build the SRTP master key. It is
- * generated by DTLS and has the following layout:
- * 16B 16B 14B 14B
- * client_key | server_key | client_salt | server_salt
- */
- uint8_t dtls_srtp_materials[(DTLS_SRTP_KEY_LEN + DTLS_SRTP_SALT_LEN) * 2];
-
- char ssl_error_message[256];
-
- /* TODO: Use AVIOContext instead of URLContext */
- URLContext *dtls_uc;
-
- /* The SRTP send context, to encrypt outgoing packets. */
- SRTPContext srtp_audio_send;
- SRTPContext srtp_video_send;
- SRTPContext srtp_video_rtx_send;
- SRTPContext srtp_rtcp_send;
- /* The SRTP receive context, to decrypt incoming packets. */
- SRTPContext srtp_recv;
-
- /* The UDP transport is used for delivering ICE, DTLS and SRTP packets. */
- URLContext *udp;
- /* The buffer for UDP transmission. */
- char buf[MAX_UDP_BUFFER_SIZE];
-
- /* The timeout in milliseconds for ICE and DTLS handshake. */
- int handshake_timeout;
- /**
- * The size of RTP packet, should generally be set to MTU.
- * Note that pion requires a smaller value, for example, 1200.
- */
- int pkt_size;
- int buffer_size;/* Underlying protocol send/receive buffer size */
- /**
- * The optional Bearer token for WHIP Authorization.
- * See https://www.ietf.org/archive/id/draft-ietf-wish-whip-08.html#name-authentication-and-authoriz
- */
- char* authorization;
- /* The certificate and private key used for DTLS handshake. */
- char* cert_file;
- char* key_file;
-} WHIPContext;
-
/**
* Whether the packet is a DTLS packet.
*/
-static int is_dtls_packet(uint8_t *b, int size) {
+int ff_rtc_is_dtls_packet(uint8_t *b, int size) {
uint16_t version = AV_RB16(&b[1]);
return size > DTLS_RECORD_LAYER_HEADER_LEN &&
b[0] >= DTLS_CONTENT_TYPE_CHANGE_CIPHER_SPEC &&
(version == DTLS_VERSION_10 || version == DTLS_VERSION_12);
}
-
/**
* Get or Generate a self-signed certificate and private key for DTLS,
* fingerprint for SDP
@@ -390,7 +187,7 @@ static av_cold int dtls_initialize(AVFormatContext *s)
/**
* Initialize and check the options for the WebRTC muxer.
*/
-static av_cold int initialize(AVFormatContext *s)
+av_cold int ff_rtc_initialize(AVFormatContext *s)
{
int ret, ideal_pkt_size = 532;
WHIPContext *whip = s->priv_data;
@@ -431,160 +228,6 @@ static av_cold int initialize(AVFormatContext *s)
return 0;
}
-/**
- * When duplicating a stream, the demuxer has already set the extradata, profile, and
- * level of the par. Keep in mind that this function will not be invoked since the
- * profile and level are set.
- *
- * When utilizing an encoder, such as libx264, to encode a stream, the extradata in
- * par->extradata contains the SPS, which includes profile and level information.
- * However, the profile and level of par remain unspecified. Therefore, it is necessary
- * to extract the profile and level data from the extradata and assign it to the par's
- * profile and level. Keep in mind that AVFMT_GLOBALHEADER must be enabled; otherwise,
- * the extradata will remain empty.
- */
-static int parse_profile_level(AVFormatContext *s, AVCodecParameters *par)
-{
- int ret = 0;
- const uint8_t *r = par->extradata, *r1, *end = par->extradata + par->extradata_size;
- H264SPS seq, *const sps = &seq;
- uint32_t state;
- WHIPContext *whip = s->priv_data;
-
- if (par->codec_id != AV_CODEC_ID_H264)
- return ret;
-
- if (par->profile != AV_PROFILE_UNKNOWN && par->level != AV_LEVEL_UNKNOWN)
- return ret;
-
- if (!par->extradata || par->extradata_size <= 0) {
- av_log(whip, AV_LOG_ERROR, "Unable to parse profile from empty extradata=%p, size=%d\n",
- par->extradata, par->extradata_size);
- return AVERROR(EINVAL);
- }
-
- while (1) {
- r = avpriv_find_start_code(r, end, &state);
- if (r >= end)
- break;
-
- r1 = ff_nal_find_startcode(r, end);
- if ((state & 0x1f) == H264_NAL_SPS) {
- ret = ff_avc_decode_sps(sps, r, r1 - r);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to decode SPS, state=%x, size=%d\n",
- state, (int)(r1 - r));
- return ret;
- }
-
- av_log(whip, AV_LOG_VERBOSE, "Parse profile=%d, level=%d from SPS\n",
- sps->profile_idc, sps->level_idc);
- par->profile = sps->profile_idc;
- par->level = sps->level_idc;
- }
-
- r = r1;
- }
-
- return ret;
-}
-
-/**
- * Parses video SPS/PPS from the extradata of codecpar and checks the codec.
- * Currently only supports video(h264) and audio(opus). Note that only baseline
- * and constrained baseline profiles of h264 are supported.
- *
- * If the profile is less than 0, the function considers the profile as baseline.
- * It may need to parse the profile from SPS/PPS. This situation occurs when ingesting
- * desktop and transcoding.
- *
- * @param s Pointer to the AVFormatContext
- * @returns Returns 0 if successful or AVERROR_xxx in case of an error.
- *
- * TODO: FIXME: There is an issue with the timestamp of OPUS audio, especially when
- * the input is an MP4 file. The timestamp deviates from the expected value of 960,
- * causing Chrome to play the audio stream with noise. This problem can be replicated
- * by transcoding a specific file into MP4 format and publishing it using the WHIP
- * muxer. However, when directly transcoding and publishing through the WHIP muxer,
- * the issue is not present, and the audio timestamp remains consistent. The root
- * cause is still unknown, and this comment has been added to address this issue
- * in the future. Further research is needed to resolve the problem.
- */
-static int parse_codec(AVFormatContext *s)
-{
- int i, ret = 0;
- WHIPContext *whip = s->priv_data;
-
- for (i = 0; i < s->nb_streams; i++) {
- AVCodecParameters *par = s->streams[i]->codecpar;
- const AVCodecDescriptor *desc = avcodec_descriptor_get(par->codec_id);
- switch (par->codec_type) {
- case AVMEDIA_TYPE_VIDEO:
- if (whip->video_par) {
- av_log(whip, AV_LOG_ERROR, "Only one video stream is supported by RTC\n");
- return AVERROR(EINVAL);
- }
- whip->video_par = par;
-
- if (par->codec_id != AV_CODEC_ID_H264) {
- av_log(whip, AV_LOG_ERROR, "Unsupported video codec %s by RTC, choose h264\n",
- desc ? desc->name : "unknown");
- return AVERROR_PATCHWELCOME;
- }
-
- if (par->video_delay > 0) {
- av_log(whip, AV_LOG_ERROR, "Unsupported B frames by RTC\n");
- return AVERROR_PATCHWELCOME;
- }
-
- if ((ret = parse_profile_level(s, par)) < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to parse SPS/PPS from extradata\n");
- return AVERROR(EINVAL);
- }
-
- if (par->profile == AV_PROFILE_UNKNOWN) {
- av_log(whip, AV_LOG_WARNING, "No profile found in extradata, consider baseline\n");
- return AVERROR(EINVAL);
- }
- if (par->level == AV_LEVEL_UNKNOWN) {
- av_log(whip, AV_LOG_WARNING, "No level found in extradata, consider 3.1\n");
- return AVERROR(EINVAL);
- }
- break;
- case AVMEDIA_TYPE_AUDIO:
- if (whip->audio_par) {
- av_log(whip, AV_LOG_ERROR, "Only one audio stream is supported by RTC\n");
- return AVERROR(EINVAL);
- }
- whip->audio_par = par;
-
- if (par->codec_id != AV_CODEC_ID_OPUS) {
- av_log(whip, AV_LOG_ERROR, "Unsupported audio codec %s by RTC, choose opus\n",
- desc ? desc->name : "unknown");
- return AVERROR_PATCHWELCOME;
- }
-
- if (par->ch_layout.nb_channels != 2) {
- av_log(whip, AV_LOG_ERROR, "Unsupported audio channels %d by RTC, choose stereo\n",
- par->ch_layout.nb_channels);
- return AVERROR_PATCHWELCOME;
- }
-
- if (par->sample_rate != 48000) {
- av_log(whip, AV_LOG_ERROR, "Unsupported audio sample rate %d by RTC, choose 48000\n", par->sample_rate);
- return AVERROR_PATCHWELCOME;
- }
- break;
- default:
- av_log(whip, AV_LOG_ERROR, "Codec type '%s' for stream %d is not supported by RTC\n",
- av_get_media_type_string(par->codec_type), i);
- return AVERROR_PATCHWELCOME;
- }
- }
-
- return ret;
-}
-
/**
* Generate SDP offer according to the codec parameters, DTLS and ICE information.
*
@@ -969,7 +612,7 @@ end:
* @param request_size Pointer to an integer that receives the size of the request packet
* @return Returns 0 if successful or AVERROR_xxx if an error occurs.
*/
-static int ice_create_request(AVFormatContext *s, uint8_t *buf, int buf_size, int *request_size)
+int ff_rtc_ice_create_request(AVFormatContext *s, uint8_t *buf, int buf_size, int *request_size)
{
int ret, size, crc32;
char username[128];
@@ -1131,7 +774,7 @@ end:
* and is encoded into the first 16 bits as 0x0001.
* See https://datatracker.ietf.org/doc/html/rfc5389#section-6
*/
-static int ice_is_binding_request(uint8_t *b, int size)
+int ff_rtc_ice_is_binding_request(uint8_t *b, int size)
{
return size >= ICE_STUN_HEADER_SIZE && AV_RB16(&b[0]) == 0x0001;
}
@@ -1140,29 +783,11 @@ static int ice_is_binding_request(uint8_t *b, int size)
* A Binding response has class=0b10 (success response) and method=0b000000000001,
* and is encoded into the first 16 bits as 0x0101.
*/
-static int ice_is_binding_response(uint8_t *b, int size)
+int ff_rtc_ice_is_binding_response(uint8_t *b, int size)
{
return size >= ICE_STUN_HEADER_SIZE && AV_RB16(&b[0]) == 0x0101;
}
-/**
- * In RTP packets, the first byte is represented as 0b10xxxxxx, where the initial
- * two bits (0b10) indicate the RTP version,
- * see https://www.rfc-editor.org/rfc/rfc3550#section-5.1
- * The RTCP packet header is similar to RTP,
- * see https://www.rfc-editor.org/rfc/rfc3550#section-6.4.1
- */
-static int media_is_rtp_rtcp(const uint8_t *b, int size)
-{
- return size >= WHIP_RTP_HEADER_SIZE && (b[0] & 0xC0) == 0x80;
-}
-
-/* Whether the packet is RTCP. */
-static int media_is_rtcp(const uint8_t *b, int size)
-{
- return size >= WHIP_RTP_HEADER_SIZE && b[1] >= WHIP_RTCP_PT_START && b[1] <= WHIP_RTCP_PT_END;
-}
-
/**
* This function handles incoming binding request messages by responding to them.
* If the message is not a binding request, it will be ignored.
@@ -1174,7 +799,7 @@ static int ice_handle_binding_request(AVFormatContext *s, char *buf, int buf_siz
WHIPContext *whip = s->priv_data;
/* Ignore if not a binding request. */
- if (!ice_is_binding_request(buf, buf_size))
+ if (!ff_rtc_ice_is_binding_request(buf, buf_size))
return ret;
if (buf_size < ICE_STUN_HEADER_SIZE) {
@@ -1261,7 +886,7 @@ static int ice_dtls_handshake(AVFormatContext *s)
while (1) {
if (whip->state <= WHIP_STATE_ICE_CONNECTING) {
/* Build the STUN binding request. */
- ret = ice_create_request(s, whip->buf, sizeof(whip->buf), &size);
+ ret = ff_rtc_ice_create_request(s, whip->buf, sizeof(whip->buf), &size);
if (ret < 0) {
av_log(whip, AV_LOG_ERROR, "Failed to create STUN binding request, size=%d\n", size);
goto end;
@@ -1306,7 +931,7 @@ next_packet:
}
/* Handle the ICE binding response. */
- if (ice_is_binding_response(whip->buf, ret)) {
+ if (ff_rtc_ice_is_binding_response(whip->buf, ret)) {
if (whip->state < WHIP_STATE_ICE_CONNECTED) {
if (whip->is_peer_ice_lite)
whip->state = WHIP_STATE_ICE_CONNECTED;
@@ -1341,14 +966,14 @@ next_packet:
}
/* When a binding request is received, it is necessary to respond immediately. */
- if (ice_is_binding_request(whip->buf, ret)) {
+ if (ff_rtc_ice_is_binding_request(whip->buf, ret)) {
if ((ret = ice_handle_binding_request(s, whip->buf, ret)) < 0)
goto end;
goto next_packet;
}
/* If got any DTLS messages, handle it. */
- if (is_dtls_packet(whip->buf, ret)) {
+ if (ff_rtc_is_dtls_packet(whip->buf, ret)) {
/* Start consent timer when ICE selected */
whip->whip_last_consent_tx_time = whip->whip_last_consent_rx_time = av_gettime_relative();
whip->state = WHIP_STATE_ICE_CONNECTED;
@@ -1473,174 +1098,6 @@ end:
return ret;
}
-/**
- * Callback triggered by the RTP muxer when it creates and sends out an RTP packet.
- *
- * This function modifies the video STAP packet, removing the markers, and updating the
- * NRI of the first NALU. Additionally, it uses the corresponding SRTP context to encrypt
- * the RTP packet, where the video packet is handled by the video SRTP context.
- */
-static int on_rtp_write_packet(void *opaque, const uint8_t *buf, int buf_size)
-{
- int ret, cipher_size, is_rtcp, is_video;
- uint8_t payload_type;
- AVFormatContext *s = opaque;
- WHIPContext *whip = s->priv_data;
- SRTPContext *srtp;
-
- /* Ignore if not RTP or RTCP packet. */
- if (!media_is_rtp_rtcp(buf, buf_size))
- return 0;
-
- /* Only support audio, video and rtcp. */
- is_rtcp = media_is_rtcp(buf, buf_size);
- payload_type = buf[1] & 0x7f;
- is_video = payload_type == whip->video_payload_type;
- if (!is_rtcp && payload_type != whip->video_payload_type && payload_type != whip->audio_payload_type)
- return 0;
-
- /* Get the corresponding SRTP context. */
- srtp = is_rtcp ? &whip->srtp_rtcp_send : (is_video? &whip->srtp_video_send : &whip->srtp_audio_send);
-
- /* Encrypt by SRTP and send out. */
- cipher_size = ff_srtp_encrypt(srtp, buf, buf_size, whip->buf, sizeof(whip->buf));
- if (cipher_size <= 0 || cipher_size < buf_size) {
- av_log(whip, AV_LOG_WARNING, "Failed to encrypt packet=%dB, cipher=%dB\n", buf_size, cipher_size);
- return 0;
- }
-
- ret = ffurl_write(whip->udp, whip->buf, cipher_size);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to write packet=%dB, ret=%d\n", cipher_size, ret);
- return ret;
- }
-
- return ret;
-}
-
-/**
- * Creates dedicated RTP muxers for each stream in the AVFormatContext to build RTP
- * packets from the encoded frames.
- *
- * The corresponding SRTP context is utilized to encrypt each stream's RTP packets. For
- * example, a video SRTP context is used for the video stream. Additionally, the
- * "on_rtp_write_packet" callback function is set as the write function for each RTP
- * muxer to send out encrypted RTP packets.
- *
- * @return 0 if OK, AVERROR_xxx on error
- */
-static int create_rtp_muxer(AVFormatContext *s)
-{
- int ret, i, is_video, buffer_size, max_packet_size;
- AVFormatContext *rtp_ctx = NULL;
- AVDictionary *opts = NULL;
- uint8_t *buffer = NULL;
- char buf[64];
- WHIPContext *whip = s->priv_data;
- whip->udp->flags |= AVIO_FLAG_NONBLOCK;
-
- const AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);
- if (!rtp_format) {
- av_log(whip, AV_LOG_ERROR, "Failed to guess rtp muxer\n");
- ret = AVERROR(ENOSYS);
- goto end;
- }
-
- /* The UDP buffer size, may greater than MTU. */
- buffer_size = MAX_UDP_BUFFER_SIZE;
- /* The RTP payload max size. Reserved some bytes for SRTP checksum and padding. */
- max_packet_size = whip->pkt_size - DTLS_SRTP_CHECKSUM_LEN;
-
- for (i = 0; i < s->nb_streams; i++) {
- rtp_ctx = avformat_alloc_context();
- if (!rtp_ctx) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
-
- rtp_ctx->oformat = rtp_format;
- if (!avformat_new_stream(rtp_ctx, NULL)) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
- /* Pass the interrupt callback on */
- rtp_ctx->interrupt_callback = s->interrupt_callback;
- /* Copy the max delay setting; the rtp muxer reads this. */
- rtp_ctx->max_delay = s->max_delay;
- /* Copy other stream parameters. */
- rtp_ctx->streams[0]->sample_aspect_ratio = s->streams[i]->sample_aspect_ratio;
- rtp_ctx->flags |= s->flags & AVFMT_FLAG_BITEXACT;
- rtp_ctx->strict_std_compliance = s->strict_std_compliance;
-
- /* Set the synchronized start time. */
- rtp_ctx->start_time_realtime = s->start_time_realtime;
-
- avcodec_parameters_copy(rtp_ctx->streams[0]->codecpar, s->streams[i]->codecpar);
- rtp_ctx->streams[0]->time_base = s->streams[i]->time_base;
-
- /**
- * For H.264, consistently utilize the annexb format through the Bitstream Filter (BSF);
- * therefore, we deactivate the extradata detection for the RTP muxer.
- */
- if (s->streams[i]->codecpar->codec_id == AV_CODEC_ID_H264) {
- av_freep(&rtp_ctx->streams[i]->codecpar->extradata);
- rtp_ctx->streams[i]->codecpar->extradata_size = 0;
- }
-
- buffer = av_malloc(buffer_size);
- if (!buffer) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
-
- rtp_ctx->pb = avio_alloc_context(buffer, buffer_size, 1, s, NULL, on_rtp_write_packet, NULL);
- if (!rtp_ctx->pb) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
- rtp_ctx->pb->max_packet_size = max_packet_size;
- rtp_ctx->pb->av_class = &ff_avio_class;
-
- is_video = s->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO;
- snprintf(buf, sizeof(buf), "%d", is_video? whip->video_payload_type : whip->audio_payload_type);
- av_dict_set(&opts, "payload_type", buf, 0);
- snprintf(buf, sizeof(buf), "%d", is_video? whip->video_ssrc : whip->audio_ssrc);
- av_dict_set(&opts, "ssrc", buf, 0);
- av_dict_set_int(&opts, "seq", is_video ? whip->video_first_seq : whip->audio_first_seq, 0);
-
- ret = avformat_write_header(rtp_ctx, &opts);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to write rtp header\n");
- goto end;
- }
-
- ff_format_set_url(rtp_ctx, av_strdup(s->url));
- s->streams[i]->time_base = rtp_ctx->streams[0]->time_base;
- s->streams[i]->priv_data = rtp_ctx;
- rtp_ctx = NULL;
- }
-
- if (whip->state < WHIP_STATE_READY)
- whip->state = WHIP_STATE_READY;
- av_log(whip, AV_LOG_INFO, "Muxer state=%d, buffer_size=%d, max_packet_size=%d, "
- "elapsed=%.2fms(init:%.2f,offer:%.2f,answer:%.2f,udp:%.2f,ice:%.2f,dtls:%.2f,srtp:%.2f)\n",
- whip->state, buffer_size, max_packet_size, ELAPSED(whip->whip_starttime, av_gettime_relative()),
- ELAPSED(whip->whip_starttime, whip->whip_init_time),
- ELAPSED(whip->whip_init_time, whip->whip_offer_time),
- ELAPSED(whip->whip_offer_time, whip->whip_answer_time),
- ELAPSED(whip->whip_answer_time, whip->whip_udp_time),
- ELAPSED(whip->whip_udp_time, whip->whip_ice_time),
- ELAPSED(whip->whip_ice_time, whip->whip_dtls_time),
- ELAPSED(whip->whip_dtls_time, whip->whip_srtp_time));
-
-end:
- if (rtp_ctx)
- avio_context_free(&rtp_ctx->pb);
- avformat_free_context(rtp_ctx);
- av_dict_free(&opts);
- return ret;
-}
-
/**
* RTC is connectionless, for it's based on UDP, so it check whether sesison is
* timeout. In such case, publishers can't republish the stream util the session
@@ -1699,98 +1156,8 @@ end:
return ret;
}
-/**
- * Since the h264_mp4toannexb filter only processes the MP4 ISOM format and bypasses
- * the annexb format, it is necessary to manually insert encoder metadata before each
- * IDR when dealing with annexb format packets. For instance, in the case of H.264,
- * we must insert SPS and PPS before the IDR frame.
- */
-static int h264_annexb_insert_sps_pps(AVFormatContext *s, AVPacket *pkt)
-{
+int ff_rtc_connect(AVFormatContext *s) {
int ret = 0;
- AVPacket *in = NULL;
- AVCodecParameters *par = s->streams[pkt->stream_index]->codecpar;
- uint32_t nal_size = 0, out_size = par ? par->extradata_size : 0;
- uint8_t unit_type, sps_seen = 0, pps_seen = 0, idr_seen = 0, *out;
- const uint8_t *buf, *buf_end, *r1;
-
- if (!par || !par->extradata || par->extradata_size <= 0)
- return ret;
-
- /* Discover NALU type from packet. */
- buf_end = pkt->data + pkt->size;
- for (buf = ff_nal_find_startcode(pkt->data, buf_end); buf < buf_end; buf += nal_size) {
- while (!*(buf++));
- r1 = ff_nal_find_startcode(buf, buf_end);
- if ((nal_size = r1 - buf) > 0) {
- unit_type = *buf & 0x1f;
- if (unit_type == H264_NAL_SPS) {
- sps_seen = 1;
- } else if (unit_type == H264_NAL_PPS) {
- pps_seen = 1;
- } else if (unit_type == H264_NAL_IDR_SLICE) {
- idr_seen = 1;
- }
-
- out_size += 3 + nal_size;
- }
- }
-
- if (!idr_seen || (sps_seen && pps_seen))
- return ret;
-
- /* See av_bsf_send_packet */
- in = av_packet_alloc();
- if (!in)
- return AVERROR(ENOMEM);
-
- ret = av_packet_make_refcounted(pkt);
- if (ret < 0)
- goto fail;
-
- av_packet_move_ref(in, pkt);
-
- /* Create a new packet with sps/pps inserted. */
- ret = av_new_packet(pkt, out_size);
- if (ret < 0)
- goto fail;
-
- ret = av_packet_copy_props(pkt, in);
- if (ret < 0)
- goto fail;
-
- memcpy(pkt->data, par->extradata, par->extradata_size);
- out = pkt->data + par->extradata_size;
- buf_end = in->data + in->size;
- for (buf = ff_nal_find_startcode(in->data, buf_end); buf < buf_end; buf += nal_size) {
- while (!*(buf++));
- r1 = ff_nal_find_startcode(buf, buf_end);
- if ((nal_size = r1 - buf) > 0) {
- AV_WB24(out, 0x00001);
- memcpy(out + 3, buf, nal_size);
- out += 3 + nal_size;
- }
- }
-
-fail:
- if (ret < 0)
- av_packet_unref(pkt);
- av_packet_free(&in);
-
- return ret;
-}
-
-static av_cold int whip_init(AVFormatContext *s)
-{
- int ret;
- WHIPContext *whip = s->priv_data;
-
- if ((ret = initialize(s)) < 0)
- goto end;
-
- if ((ret = parse_codec(s)) < 0)
- goto end;
-
if ((ret = generate_sdp_offer(s)) < 0)
goto end;
@@ -1809,152 +1176,11 @@ static av_cold int whip_init(AVFormatContext *s)
if ((ret = setup_srtp(s)) < 0)
goto end;
- if ((ret = create_rtp_muxer(s)) < 0)
- goto end;
-
-end:
- if (ret < 0)
- whip->state = WHIP_STATE_FAILED;
- return ret;
-}
-
-static void handle_nack_rtx(AVFormatContext *s, int size)
-{
- int ret;
- WHIPContext *whip = s->priv_data;
- uint8_t *buf = NULL;
- int rtcp_len, srtcp_len, header_len = 12/*RFC 4585 6.1*/;
-
- /**
- * Refer to RFC 3550 6.4.1
- * The length of this RTCP packet in 32 bit words minus one,
- * including the header and any padding.
- */
- rtcp_len = (AV_RB16(&whip->buf[2]) + 1) * 4;
- if (rtcp_len <= header_len) {
- av_log(whip, AV_LOG_WARNING, "NACK packet is broken, size: %d\n", rtcp_len);
- goto error;
- }
- /* SRTCP index(4 bytes) + HMAC(SRTP_ARS128_CM_SHA1_80) 10bytes */
- srtcp_len = rtcp_len + 4 + 10;
- if (srtcp_len != size) {
- av_log(whip, AV_LOG_WARNING, "NACK packet size not match, srtcp_len:%d, size:%d\n", srtcp_len, size);
- goto error;
- }
- buf = av_memdup(whip->buf, srtcp_len);
- if (!buf)
- goto error;
- if ((ret = ff_srtp_decrypt(&whip->srtp_recv, buf, &srtcp_len)) < 0) {
- av_log(whip, AV_LOG_WARNING, "NACK packet decrypt failed: %d\n", ret);
- goto error;
- }
- goto end;
-error:
- av_log(whip, AV_LOG_WARNING, "Failed to handle NACK and RTX, Skip...\n");
-end:
- av_freep(&buf);
-}
-
-static int whip_write_packet(AVFormatContext *s, AVPacket *pkt)
-{
- int ret;
- WHIPContext *whip = s->priv_data;
- AVStream *st = s->streams[pkt->stream_index];
- AVFormatContext *rtp_ctx = st->priv_data;
- int64_t now = av_gettime_relative();
- /**
- * Refer to RFC 7675
- * Periodically send Consent Freshness STUN Binding Request
- */
- if (now - whip->whip_last_consent_tx_time > WHIP_ICE_CONSENT_CHECK_INTERVAL * WHIP_US_PER_MS) {
- int size;
- ret = ice_create_request(s, whip->buf, sizeof(whip->buf), &size);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to create STUN binding request, size=%d\n", size);
- goto end;
- }
- ret = ffurl_write(whip->udp, whip->buf, size);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to send STUN binding request, size=%d\n", size);
- goto end;
- }
- whip->whip_last_consent_tx_time = now;
- av_log(whip, AV_LOG_DEBUG, "Consent Freshness check sent\n");
- }
-
- /**
- * Receive packets from the server such as ICE binding requests, DTLS messages,
- * and RTCP like PLI requests, then respond to them.
- */
- ret = ffurl_read(whip->udp, whip->buf, sizeof(whip->buf));
- if (ret < 0) {
- if (ret == AVERROR(EAGAIN))
- goto write_packet;
- av_log(whip, AV_LOG_ERROR, "Failed to read from UDP socket\n");
- goto end;
- }
- if (!ret) {
- av_log(whip, AV_LOG_ERROR, "Receive EOF from UDP socket\n");
- goto end;
- }
- if (ice_is_binding_response(whip->buf, ret)) {
- whip->whip_last_consent_rx_time = av_gettime_relative();
- av_log(whip, AV_LOG_DEBUG, "Consent Freshness check received\n");
- }
- if (is_dtls_packet(whip->buf, ret)) {
- if ((ret = ffurl_write(whip->dtls_uc, whip->buf, ret)) < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to handle DTLS message\n");
- goto end;
- }
- }
- if (media_is_rtcp(whip->buf, ret)) {
- uint8_t fmt = whip->buf[0] & 0x1f;
- uint8_t pt = whip->buf[1];
- /**
- * Handle RTCP NACK packet
- * Refer to RFC 4585 6.2.1
- * The Generic NACK message is identified by PT=RTPFB and FMT=1
- */
- if (pt != RTCP_RTPFB)
- goto write_packet;
- if (fmt == 1)
- handle_nack_rtx(s, ret);
- }
-write_packet:
- now = av_gettime_relative();
- if (now - whip->whip_last_consent_rx_time > WHIP_ICE_CONSENT_EXPIRED_TIMER * WHIP_US_PER_MS) {
- av_log(whip, AV_LOG_ERROR,
- "Consent Freshness expired after %.2fms (limited %dms), terminate session\n",
- ELAPSED(now, whip->whip_last_consent_rx_time), WHIP_ICE_CONSENT_EXPIRED_TIMER);
- ret = AVERROR(ETIMEDOUT);
- goto end;
- }
- if (whip->h264_annexb_insert_sps_pps && st->codecpar->codec_id == AV_CODEC_ID_H264) {
- if ((ret = h264_annexb_insert_sps_pps(s, pkt)) < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to insert SPS/PPS before IDR\n");
- goto end;
- }
- }
-
- ret = ff_write_chained(rtp_ctx, 0, pkt, s, 0);
- if (ret < 0) {
- if (ret == AVERROR(EINVAL)) {
- av_log(whip, AV_LOG_WARNING, "Ignore failed to write packet=%dB, ret=%d\n", pkt->size, ret);
- ret = 0;
- } else if (ret == AVERROR(EAGAIN)) {
- av_log(whip, AV_LOG_ERROR, "UDP send blocked, please increase the buffer via -buffer_size\n");
- } else
- av_log(whip, AV_LOG_ERROR, "Failed to write packet, size=%d, ret=%d\n", pkt->size, ret);
- goto end;
- }
-
end:
- if (ret < 0)
- whip->state = WHIP_STATE_FAILED;
return ret;
}
-static av_cold void whip_deinit(AVFormatContext *s)
+void ff_rtc_close(AVFormatContext *s)
{
int i, ret;
WHIPContext *whip = s->priv_data;
@@ -1999,28 +1225,9 @@ static av_cold void whip_deinit(AVFormatContext *s)
ffurl_closep(&whip->udp);
}
-static int whip_check_bitstream(AVFormatContext *s, AVStream *st, const AVPacket *pkt)
-{
- int ret = 1, extradata_isom = 0;
- uint8_t *b = pkt->data;
- WHIPContext *whip = s->priv_data;
-
- if (st->codecpar->codec_id == AV_CODEC_ID_H264) {
- extradata_isom = st->codecpar->extradata_size > 0 && st->codecpar->extradata[0] == 1;
- if (pkt->size >= 5 && AV_RB32(b) != 0x0000001 && (AV_RB24(b) != 0x000001 || extradata_isom)) {
- ret = ff_stream_add_bitstream_filter(st, "h264_mp4toannexb", NULL);
- av_log(whip, AV_LOG_VERBOSE, "Enable BSF h264_mp4toannexb, packet=[%x %x %x %x %x ...], extradata_isom=%d\n",
- b[0], b[1], b[2], b[3], b[4], extradata_isom);
- } else
- whip->h264_annexb_insert_sps_pps = 1;
- }
-
- return ret;
-}
-
#define OFFSET(x) offsetof(WHIPContext, x)
#define ENC AV_OPT_FLAG_ENCODING_PARAM
-static const AVOption options[] = {
+const AVOption ff_rtc_options[] = {
{ "handshake_timeout", "Timeout in milliseconds for ICE and DTLS handshake.", OFFSET(handshake_timeout), AV_OPT_TYPE_INT, { .i64 = 5000 }, -1, INT_MAX, ENC },
{ "pkt_size", "The maximum size, in bytes, of RTP packets that send out", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = 1200 }, -1, INT_MAX, ENC },
{ "buffer_size", "The buffer size, in bytes, of underlying protocol", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC },
@@ -2029,24 +1236,3 @@ static const AVOption options[] = {
{ "key_file", "The optional private key file path for DTLS", OFFSET(key_file), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, ENC },
{ NULL },
};
-
-static const AVClass whip_muxer_class = {
- .class_name = "WHIP muxer",
- .item_name = av_default_item_name,
- .option = options,
- .version = LIBAVUTIL_VERSION_INT,
-};
-
-const FFOutputFormat ff_whip_muxer = {
- .p.name = "whip",
- .p.long_name = NULL_IF_CONFIG_SMALL("WHIP(WebRTC-HTTP ingestion protocol) muxer"),
- .p.audio_codec = AV_CODEC_ID_OPUS,
- .p.video_codec = AV_CODEC_ID_H264,
- .p.flags = AVFMT_GLOBALHEADER | AVFMT_NOFILE | AVFMT_EXPERIMENTAL,
- .p.priv_class = &whip_muxer_class,
- .priv_data_size = sizeof(WHIPContext),
- .init = whip_init,
- .write_packet = whip_write_packet,
- .deinit = whip_deinit,
- .check_bitstream = whip_check_bitstream,
-};
diff --git a/libavformat/rtc.h b/libavformat/rtc.h
new file mode 100644
index 0000000000..146ad06f31
--- /dev/null
+++ b/libavformat/rtc.h
@@ -0,0 +1,220 @@
+/*
+ * RTC definitions
+ * Copyright (c) 2002 Fabrice Bellard
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVFORMAT_RTC_H
+#define AVFORMAT_RTC_H
+
+#include <stdint.h>
+#include "avformat.h"
+#include "url.h"
+#include "tls.h"
+#include "srtp.h"
+
+#include "libavutil/lfg.h"
+#include "libavutil/log.h"
+#include "libavutil/opt.h"
+
+enum WHIPState {
+ WHIP_STATE_NONE,
+
+ /* The initial state. */
+ WHIP_STATE_INIT,
+ /* The muxer has sent the offer to the peer. */
+ WHIP_STATE_OFFER,
+ /* The muxer has received the answer from the peer. */
+ WHIP_STATE_ANSWER,
+ /**
+ * After parsing the answer received from the peer, the muxer negotiates the abilities
+ * in the offer that it generated.
+ */
+ WHIP_STATE_NEGOTIATED,
+ /* The muxer has connected to the peer via UDP. */
+ WHIP_STATE_UDP_CONNECTED,
+ /* The muxer has sent the ICE request to the peer. */
+ WHIP_STATE_ICE_CONNECTING,
+ /* The muxer has received the ICE response from the peer. */
+ WHIP_STATE_ICE_CONNECTED,
+ /* The muxer has finished the DTLS handshake with the peer. */
+ WHIP_STATE_DTLS_FINISHED,
+ /* The muxer has finished the SRTP setup. */
+ WHIP_STATE_SRTP_FINISHED,
+ /* The muxer is ready to send/receive media frames. */
+ WHIP_STATE_READY,
+ /* The muxer is failed. */
+ WHIP_STATE_FAILED,
+};
+
+/**
+ * The size of the Secure Real-time Transport Protocol (SRTP) master key material
+ * that is exported by Secure Sockets Layer (SSL) after a successful Datagram
+ * Transport Layer Security (DTLS) handshake. This material consists of a key
+ * of 16 bytes and a salt of 14 bytes.
+ */
+#define DTLS_SRTP_KEY_LEN 16
+#define DTLS_SRTP_SALT_LEN 14
+#define WHIP_US_PER_MS 1000
+
+/**
+ * Maximum size of the buffer for sending and receiving UDP packets.
+ * Please note that this size does not limit the size of the UDP packet that can be sent.
+ * To set the limit for packet size, modify the `pkt_size` parameter.
+ * For instance, it is possible to set the UDP buffer to 4096 to send or receive packets,
+ * but please keep in mind that the `pkt_size` option limits the packet size to 1400.
+ */
+#define MAX_UDP_BUFFER_SIZE 4096
+
+typedef struct WHIPContext {
+ AVClass *av_class;
+
+ /* The state of the RTC connection. */
+ enum WHIPState state;
+
+ /* Parameters for the input audio and video codecs. */
+ AVCodecParameters *audio_par;
+ AVCodecParameters *video_par;
+
+ /**
+ * The h264_mp4toannexb Bitstream Filter (BSF) bypasses the AnnexB packet;
+ * therefore, it is essential to insert the SPS and PPS before each IDR frame
+ * in such cases.
+ */
+ int h264_annexb_insert_sps_pps;
+
+ /* The random number generator. */
+ AVLFG rnd;
+
+ /* The ICE username and pwd fragment generated by the muxer. */
+ char ice_ufrag_local[9];
+ char ice_pwd_local[33];
+ /* The SSRC of the audio and video stream, generated by the muxer. */
+ uint32_t audio_ssrc;
+ uint32_t video_ssrc;
+ uint32_t video_rtx_ssrc;
+
+ uint16_t audio_first_seq;
+ uint16_t video_first_seq;
+ /* The PT(Payload Type) of stream, generated by the muxer. */
+ uint8_t audio_payload_type;
+ uint8_t video_payload_type;
+ uint8_t video_rtx_payload_type;
+ /**
+ * This is the SDP offer generated by the muxer based on the codec parameters,
+ * DTLS, and ICE information.
+ */
+ char *sdp_offer;
+
+ int is_peer_ice_lite;
+ uint64_t ice_tie_breaker; // random 64 bit, for ICE-CONTROLLING
+ /* The ICE username and pwd from remote server. */
+ char *ice_ufrag_remote;
+ char *ice_pwd_remote;
+ /**
+ * This represents the ICE candidate protocol, priority, host and port.
+ * Currently, we only support one candidate and choose the first UDP candidate.
+ * However, we plan to support multiple candidates in the future.
+ */
+ char *ice_protocol;
+ char *ice_host;
+ int ice_port;
+
+ /* The SDP answer received from the WebRTC server. */
+ char *sdp_answer;
+ /* The resource URL returned in the Location header of WHIP HTTP response. */
+ char *whip_resource_url;
+
+ /* These variables represent timestamps used for calculating and tracking the cost. */
+ int64_t whip_starttime;
+ int64_t whip_init_time;
+ int64_t whip_offer_time;
+ int64_t whip_answer_time;
+ int64_t whip_udp_time;
+ int64_t whip_ice_time;
+ int64_t whip_dtls_time;
+ int64_t whip_srtp_time;
+ int64_t whip_last_consent_tx_time;
+ int64_t whip_last_consent_rx_time;
+
+ /* The certificate and private key content used for DTLS handshake */
+ char cert_buf[MAX_CERTIFICATE_SIZE];
+ char key_buf[MAX_CERTIFICATE_SIZE];
+ /* The fingerprint of certificate, used in SDP offer. */
+ char *dtls_fingerprint;
+ /**
+ * This represents the material used to build the SRTP master key. It is
+ * generated by DTLS and has the following layout:
+ * 16B 16B 14B 14B
+ * client_key | server_key | client_salt | server_salt
+ */
+ uint8_t dtls_srtp_materials[(DTLS_SRTP_KEY_LEN + DTLS_SRTP_SALT_LEN) * 2];
+
+ char ssl_error_message[256];
+
+ /* TODO: Use AVIOContext instead of URLContext */
+ URLContext *dtls_uc;
+
+ /* The SRTP send context, to encrypt outgoing packets. */
+ SRTPContext srtp_audio_send;
+ SRTPContext srtp_video_send;
+ SRTPContext srtp_video_rtx_send;
+ SRTPContext srtp_rtcp_send;
+ /* The SRTP receive context, to decrypt incoming packets. */
+ SRTPContext srtp_recv;
+
+ /* The UDP transport is used for delivering ICE, DTLS and SRTP packets. */
+ URLContext *udp;
+ /* The buffer for UDP transmission. */
+ char buf[MAX_UDP_BUFFER_SIZE];
+
+ /* The timeout in milliseconds for ICE and DTLS handshake. */
+ int handshake_timeout;
+ /**
+ * The size of RTP packet, should generally be set to MTU.
+ * Note that pion requires a smaller value, for example, 1200.
+ */
+ int pkt_size;
+ int buffer_size;/* Underlying protocol send/receive buffer size */
+ /**
+ * The optional Bearer token for WHIP Authorization.
+ * See https://www.ietf.org/archive/id/draft-ietf-wish-whip-08.html#name-authentication-and-authoriz
+ */
+ char* authorization;
+ /* The certificate and private key used for DTLS handshake. */
+ char* cert_file;
+ char* key_file;
+} WHIPContext;
+
+int ff_rtc_initialize(AVFormatContext *s);
+
+int ff_rtc_connect(AVFormatContext *s);
+
+void ff_rtc_close(AVFormatContext *s);
+
+int ff_rtc_is_dtls_packet(uint8_t *b, int size);
+
+int ff_rtc_ice_is_binding_request(uint8_t *b, int size);
+
+int ff_rtc_ice_is_binding_response(uint8_t *b, int size);
+
+int ff_rtc_ice_create_request(AVFormatContext *s, uint8_t *buf, int buf_size, int *request_size);
+
+extern const AVOption ff_rtc_options[];
+
+#endif /* AVFORMAT_RTC_H */
diff --git a/libavformat/whip.c b/libavformat/whip.c
index e809075643..8e517f62ee 100644
--- a/libavformat/whip.c
+++ b/libavformat/whip.c
@@ -23,41 +23,19 @@
#include "libavcodec/codec_desc.h"
#include "libavcodec/h264.h"
#include "libavcodec/startcode.h"
-#include "libavutil/base64.h"
#include "libavutil/bprint.h"
-#include "libavutil/crc.h"
-#include "libavutil/hmac.h"
#include "libavutil/intreadwrite.h"
-#include "libavutil/lfg.h"
-#include "libavutil/opt.h"
#include "libavutil/mem.h"
#include "libavutil/random_seed.h"
#include "libavutil/time.h"
#include "avc.h"
#include "nal.h"
#include "avio_internal.h"
-#include "http.h"
#include "internal.h"
#include "mux.h"
-#include "network.h"
#include "rtp.h"
-#include "srtp.h"
-#include "tls.h"
+#include "rtc.h"
-/**
- * Maximum size limit of a Session Description Protocol (SDP),
- * be it an offer or answer.
- */
-#define MAX_SDP_SIZE 8192
-
-/**
- * The size of the Secure Real-time Transport Protocol (SRTP) master key material
- * that is exported by Secure Sockets Layer (SSL) after a successful Datagram
- * Transport Layer Security (DTLS) handshake. This material consists of a key
- * of 16 bytes and a salt of 14 bytes.
- */
-#define DTLS_SRTP_KEY_LEN 16
-#define DTLS_SRTP_SALT_LEN 14
/**
* The maximum size of the Secure Real-time Transport Protocol (SRTP) HMAC checksum
@@ -67,69 +45,6 @@
*/
#define DTLS_SRTP_CHECKSUM_LEN 16
-#define WHIP_US_PER_MS 1000
-
-/**
- * If we try to read from UDP and get EAGAIN, we sleep for 5ms and retry up to 10 times.
- * This will limit the total duration (in milliseconds, 50ms)
- */
-#define ICE_DTLS_READ_MAX_RETRY 10
-#define ICE_DTLS_READ_SLEEP_DURATION 5
-
-/* The magic cookie for Session Traversal Utilities for NAT (STUN) messages. */
-#define STUN_MAGIC_COOKIE 0x2112A442
-
-/**
- * Refer to RFC 8445 5.1.2
- * priority = (2^24)*(type preference) + (2^8)*(local preference) + (2^0)*(256 - component ID)
- * host candidate priority is 126 << 24 | 65535 << 8 | 255
- */
-#define STUN_HOST_CANDIDATE_PRIORITY 126 << 24 | 65535 << 8 | 255
-
-/**
- * The DTLS content type.
- * See https://tools.ietf.org/html/rfc2246#section-6.2.1
- * change_cipher_spec(20), alert(21), handshake(22), application_data(23)
- */
-#define DTLS_CONTENT_TYPE_CHANGE_CIPHER_SPEC 20
-
-/**
- * The DTLS record layer header has a total size of 13 bytes, consisting of
- * ContentType (1 byte), ProtocolVersion (2 bytes), Epoch (2 bytes),
- * SequenceNumber (6 bytes), and Length (2 bytes).
- * See https://datatracker.ietf.org/doc/html/rfc9147#section-4
- */
-#define DTLS_RECORD_LAYER_HEADER_LEN 13
-
-/**
- * The DTLS version number, which is 0xfeff for DTLS 1.0, or 0xfefd for DTLS 1.2.
- * See https://datatracker.ietf.org/doc/html/rfc9147#name-the-dtls-record-layer
- */
-#define DTLS_VERSION_10 0xfeff
-#define DTLS_VERSION_12 0xfefd
-
-/**
- * Maximum size of the buffer for sending and receiving UDP packets.
- * Please note that this size does not limit the size of the UDP packet that can be sent.
- * To set the limit for packet size, modify the `pkt_size` parameter.
- * For instance, it is possible to set the UDP buffer to 4096 to send or receive packets,
- * but please keep in mind that the `pkt_size` option limits the packet size to 1400.
- */
-#define MAX_UDP_BUFFER_SIZE 4096
-
-/* Referring to Chrome's definition of RTP payload types. */
-#define WHIP_RTP_PAYLOAD_TYPE_H264 106
-#define WHIP_RTP_PAYLOAD_TYPE_OPUS 111
-#define WHIP_RTP_PAYLOAD_TYPE_VIDEO_RTX 105
-
-/**
- * The STUN message header, which is 20 bytes long, comprises the
- * STUNMessageType (1B), MessageLength (2B), MagicCookie (4B),
- * and TransactionID (12B).
- * See https://datatracker.ietf.org/doc/html/rfc5389#section-6
- */
-#define ICE_STUN_HEADER_SIZE 20
-
/**
* The RTP header is 12 bytes long, comprising the Version(1B), PT(1B),
* SequenceNumber(2B), Timestamp(4B), and SSRC(4B).
@@ -150,13 +65,6 @@
#define WHIP_RTCP_PT_START 192
#define WHIP_RTCP_PT_END 223
-/**
- * In the case of ICE-LITE, these fields are not used; instead, they are defined
- * as constant values.
- */
-#define WHIP_SDP_SESSION_ID "4489045141692799359"
-#define WHIP_SDP_CREATOR_IP "127.0.0.1"
-
/**
* Refer to RFC 7675 5.1,
*
@@ -171,264 +79,22 @@
/* Calculate the elapsed time from starttime to endtime in milliseconds. */
#define ELAPSED(starttime, endtime) ((float)(endtime - starttime) / 1000)
-/* STUN Attribute, comprehension-required range (0x0000-0x7FFF) */
-enum STUNAttr {
- STUN_ATTR_USERNAME = 0x0006, /// shared secret response/bind request
- STUN_ATTR_PRIORITY = 0x0024, /// must be included in a Binding request
- STUN_ATTR_USE_CANDIDATE = 0x0025, /// bind request
- STUN_ATTR_MESSAGE_INTEGRITY = 0x0008, /// bind request/response
- STUN_ATTR_FINGERPRINT = 0x8028, /// rfc5389
- STUN_ATTR_ICE_CONTROLLING = 0x802A, /// ICE controlling role
-};
-
-enum WHIPState {
- WHIP_STATE_NONE,
-
- /* The initial state. */
- WHIP_STATE_INIT,
- /* The muxer has sent the offer to the peer. */
- WHIP_STATE_OFFER,
- /* The muxer has received the answer from the peer. */
- WHIP_STATE_ANSWER,
- /**
- * After parsing the answer received from the peer, the muxer negotiates the abilities
- * in the offer that it generated.
- */
- WHIP_STATE_NEGOTIATED,
- /* The muxer has connected to the peer via UDP. */
- WHIP_STATE_UDP_CONNECTED,
- /* The muxer has sent the ICE request to the peer. */
- WHIP_STATE_ICE_CONNECTING,
- /* The muxer has received the ICE response from the peer. */
- WHIP_STATE_ICE_CONNECTED,
- /* The muxer has finished the DTLS handshake with the peer. */
- WHIP_STATE_DTLS_FINISHED,
- /* The muxer has finished the SRTP setup. */
- WHIP_STATE_SRTP_FINISHED,
- /* The muxer is ready to send/receive media frames. */
- WHIP_STATE_READY,
- /* The muxer is failed. */
- WHIP_STATE_FAILED,
-};
-
-typedef struct WHIPContext {
- AVClass *av_class;
-
- /* The state of the RTC connection. */
- enum WHIPState state;
-
- /* Parameters for the input audio and video codecs. */
- AVCodecParameters *audio_par;
- AVCodecParameters *video_par;
-
- /**
- * The h264_mp4toannexb Bitstream Filter (BSF) bypasses the AnnexB packet;
- * therefore, it is essential to insert the SPS and PPS before each IDR frame
- * in such cases.
- */
- int h264_annexb_insert_sps_pps;
-
- /* The random number generator. */
- AVLFG rnd;
-
- /* The ICE username and pwd fragment generated by the muxer. */
- char ice_ufrag_local[9];
- char ice_pwd_local[33];
- /* The SSRC of the audio and video stream, generated by the muxer. */
- uint32_t audio_ssrc;
- uint32_t video_ssrc;
- uint32_t video_rtx_ssrc;
-
- uint16_t audio_first_seq;
- uint16_t video_first_seq;
- /* The PT(Payload Type) of stream, generated by the muxer. */
- uint8_t audio_payload_type;
- uint8_t video_payload_type;
- uint8_t video_rtx_payload_type;
- /**
- * This is the SDP offer generated by the muxer based on the codec parameters,
- * DTLS, and ICE information.
- */
- char *sdp_offer;
-
- int is_peer_ice_lite;
- uint64_t ice_tie_breaker; // random 64 bit, for ICE-CONTROLLING
- /* The ICE username and pwd from remote server. */
- char *ice_ufrag_remote;
- char *ice_pwd_remote;
- /**
- * This represents the ICE candidate protocol, priority, host and port.
- * Currently, we only support one candidate and choose the first UDP candidate.
- * However, we plan to support multiple candidates in the future.
- */
- char *ice_protocol;
- char *ice_host;
- int ice_port;
-
- /* The SDP answer received from the WebRTC server. */
- char *sdp_answer;
- /* The resource URL returned in the Location header of WHIP HTTP response. */
- char *whip_resource_url;
-
- /* These variables represent timestamps used for calculating and tracking the cost. */
- int64_t whip_starttime;
- int64_t whip_init_time;
- int64_t whip_offer_time;
- int64_t whip_answer_time;
- int64_t whip_udp_time;
- int64_t whip_ice_time;
- int64_t whip_dtls_time;
- int64_t whip_srtp_time;
- int64_t whip_last_consent_tx_time;
- int64_t whip_last_consent_rx_time;
-
- /* The certificate and private key content used for DTLS handshake */
- char cert_buf[MAX_CERTIFICATE_SIZE];
- char key_buf[MAX_CERTIFICATE_SIZE];
- /* The fingerprint of certificate, used in SDP offer. */
- char *dtls_fingerprint;
- /**
- * This represents the material used to build the SRTP master key. It is
- * generated by DTLS and has the following layout:
- * 16B 16B 14B 14B
- * client_key | server_key | client_salt | server_salt
- */
- uint8_t dtls_srtp_materials[(DTLS_SRTP_KEY_LEN + DTLS_SRTP_SALT_LEN) * 2];
-
- char ssl_error_message[256];
-
- /* TODO: Use AVIOContext instead of URLContext */
- URLContext *dtls_uc;
-
- /* The SRTP send context, to encrypt outgoing packets. */
- SRTPContext srtp_audio_send;
- SRTPContext srtp_video_send;
- SRTPContext srtp_video_rtx_send;
- SRTPContext srtp_rtcp_send;
- /* The SRTP receive context, to decrypt incoming packets. */
- SRTPContext srtp_recv;
-
- /* The UDP transport is used for delivering ICE, DTLS and SRTP packets. */
- URLContext *udp;
- /* The buffer for UDP transmission. */
- char buf[MAX_UDP_BUFFER_SIZE];
-
- /* The timeout in milliseconds for ICE and DTLS handshake. */
- int handshake_timeout;
- /**
- * The size of RTP packet, should generally be set to MTU.
- * Note that pion requires a smaller value, for example, 1200.
- */
- int pkt_size;
- int buffer_size;/* Underlying protocol send/receive buffer size */
- /**
- * The optional Bearer token for WHIP Authorization.
- * See https://www.ietf.org/archive/id/draft-ietf-wish-whip-08.html#name-authentication-and-authoriz
- */
- char* authorization;
- /* The certificate and private key used for DTLS handshake. */
- char* cert_file;
- char* key_file;
-} WHIPContext;
-
-/**
- * Whether the packet is a DTLS packet.
- */
-static int is_dtls_packet(uint8_t *b, int size) {
- uint16_t version = AV_RB16(&b[1]);
- return size > DTLS_RECORD_LAYER_HEADER_LEN &&
- b[0] >= DTLS_CONTENT_TYPE_CHANGE_CIPHER_SPEC &&
- (version == DTLS_VERSION_10 || version == DTLS_VERSION_12);
-}
-
-
/**
- * Get or Generate a self-signed certificate and private key for DTLS,
- * fingerprint for SDP
+ * In RTP packets, the first byte is represented as 0b10xxxxxx, where the initial
+ * two bits (0b10) indicate the RTP version,
+ * see https://www.rfc-editor.org/rfc/rfc3550#section-5.1
+ * The RTCP packet header is similar to RTP,
+ * see https://www.rfc-editor.org/rfc/rfc3550#section-6.4.1
*/
-static av_cold int certificate_key_init(AVFormatContext *s)
-{
- int ret = 0;
- WHIPContext *whip = s->priv_data;
-
- if (whip->cert_file && whip->key_file) {
- /* Read the private key and certificate from the file. */
- if ((ret = ff_ssl_read_key_cert(whip->key_file, whip->cert_file,
- whip->key_buf, sizeof(whip->key_buf),
- whip->cert_buf, sizeof(whip->cert_buf),
- &whip->dtls_fingerprint)) < 0) {
- av_log(s, AV_LOG_ERROR, "Failed to read DTLS certificate from cert=%s, key=%s\n",
- whip->cert_file, whip->key_file);
- return ret;
- }
- } else {
- /* Generate a private key to ctx->dtls_pkey and self-signed certificate. */
- if ((ret = ff_ssl_gen_key_cert(whip->key_buf, sizeof(whip->key_buf),
- whip->cert_buf, sizeof(whip->cert_buf),
- &whip->dtls_fingerprint)) < 0) {
- av_log(s, AV_LOG_ERROR, "Failed to generate DTLS private key and certificate\n");
- return ret;
- }
- }
-
- return ret;
-}
-
-static av_cold int dtls_initialize(AVFormatContext *s)
+static int media_is_rtp_rtcp(const uint8_t *b, int size)
{
- WHIPContext *whip = s->priv_data;
- /* reuse the udp created by whip */
- ff_tls_set_external_socket(whip->dtls_uc, whip->udp);
-
- /* Make the socket non-blocking */
- ff_socket_nonblock(ffurl_get_file_handle(whip->dtls_uc), 1);
- whip->dtls_uc->flags |= AVIO_FLAG_NONBLOCK;
-
- return 0;
+ return size >= WHIP_RTP_HEADER_SIZE && (b[0] & 0xC0) == 0x80;
}
-/**
- * Initialize and check the options for the WebRTC muxer.
- */
-static av_cold int initialize(AVFormatContext *s)
+/* Whether the packet is RTCP. */
+static int media_is_rtcp(const uint8_t *b, int size)
{
- int ret, ideal_pkt_size = 532;
- WHIPContext *whip = s->priv_data;
- uint32_t seed;
-
- whip->whip_starttime = av_gettime_relative();
-
- ret = certificate_key_init(s);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to init certificate and key\n");
- return ret;
- }
-
- /* Initialize the random number generator. */
- seed = av_get_random_seed();
- av_lfg_init(&whip->rnd, seed);
-
- /* 64 bit tie breaker for ICE-CONTROLLING (RFC 8445 16.1) */
- ret = av_random_bytes((uint8_t *)&whip->ice_tie_breaker, sizeof(whip->ice_tie_breaker));
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Couldn't generate random bytes for ICE tie breaker\n");
- return ret;
- }
-
- whip->audio_first_seq = av_lfg_get(&whip->rnd) & 0x0fff;
- whip->video_first_seq = whip->audio_first_seq + 1;
-
- if (whip->pkt_size < ideal_pkt_size)
- av_log(whip, AV_LOG_WARNING, "pkt_size=%d(<%d) is too small, may cause packet loss\n",
- whip->pkt_size, ideal_pkt_size);
-
- if (whip->state < WHIP_STATE_INIT)
- whip->state = WHIP_STATE_INIT;
- whip->whip_init_time = av_gettime_relative();
- av_log(whip, AV_LOG_VERBOSE, "Init state=%d, handshake_timeout=%dms, pkt_size=%d, seed=%d, elapsed=%.2fms\n",
- whip->state, whip->handshake_timeout, whip->pkt_size, seed, ELAPSED(whip->whip_starttime, av_gettime_relative()));
-
- return 0;
+ return size >= WHIP_RTP_HEADER_SIZE && b[1] >= WHIP_RTCP_PT_START && b[1] <= WHIP_RTCP_PT_END;
}
/**
@@ -585,893 +251,6 @@ static int parse_codec(AVFormatContext *s)
return ret;
}
-/**
- * Generate SDP offer according to the codec parameters, DTLS and ICE information.
- *
- * Note that we don't use av_sdp_create to generate SDP offer because it doesn't
- * support DTLS and ICE information.
- *
- * @return 0 if OK, AVERROR_xxx on error
- */
-static int generate_sdp_offer(AVFormatContext *s)
-{
- int ret = 0, profile_idc = 0, level, profile_iop = 0;
- const char *acodec_name = NULL, *vcodec_name = NULL;
- AVBPrint bp;
- WHIPContext *whip = s->priv_data;
-
- /* To prevent a crash during cleanup, always initialize it. */
- av_bprint_init(&bp, 1, MAX_SDP_SIZE);
-
- if (whip->sdp_offer) {
- av_log(whip, AV_LOG_ERROR, "SDP offer is already set\n");
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- snprintf(whip->ice_ufrag_local, sizeof(whip->ice_ufrag_local), "%08x",
- av_lfg_get(&whip->rnd));
- snprintf(whip->ice_pwd_local, sizeof(whip->ice_pwd_local), "%08x%08x%08x%08x",
- av_lfg_get(&whip->rnd), av_lfg_get(&whip->rnd), av_lfg_get(&whip->rnd),
- av_lfg_get(&whip->rnd));
-
- whip->audio_ssrc = av_lfg_get(&whip->rnd);
- whip->video_ssrc = whip->audio_ssrc + 1;
- whip->video_rtx_ssrc = whip->video_ssrc + 1;
-
- whip->audio_payload_type = WHIP_RTP_PAYLOAD_TYPE_OPUS;
- whip->video_payload_type = WHIP_RTP_PAYLOAD_TYPE_H264;
- whip->video_rtx_payload_type = WHIP_RTP_PAYLOAD_TYPE_VIDEO_RTX;
-
- av_bprintf(&bp, ""
- "v=0\r\n"
- "o=FFmpeg %s 2 IN IP4 %s\r\n"
- "s=FFmpegPublishSession\r\n"
- "t=0 0\r\n"
- "a=group:BUNDLE 0 1\r\n"
- "a=extmap-allow-mixed\r\n"
- "a=msid-semantic: WMS\r\n",
- WHIP_SDP_SESSION_ID,
- WHIP_SDP_CREATOR_IP);
-
- if (whip->audio_par) {
- if (whip->audio_par->codec_id == AV_CODEC_ID_OPUS)
- acodec_name = "opus";
-
- av_bprintf(&bp, ""
- "m=audio 9 UDP/TLS/RTP/SAVPF %u\r\n"
- "c=IN IP4 0.0.0.0\r\n"
- "a=ice-ufrag:%s\r\n"
- "a=ice-pwd:%s\r\n"
- "a=fingerprint:sha-256 %s\r\n"
- "a=setup:passive\r\n"
- "a=mid:0\r\n"
- "a=sendonly\r\n"
- "a=msid:FFmpeg audio\r\n"
- "a=rtcp-mux\r\n"
- "a=rtpmap:%u %s/%d/%d\r\n"
- "a=ssrc:%u cname:FFmpeg\r\n"
- "a=ssrc:%u msid:FFmpeg audio\r\n",
- whip->audio_payload_type,
- whip->ice_ufrag_local,
- whip->ice_pwd_local,
- whip->dtls_fingerprint,
- whip->audio_payload_type,
- acodec_name,
- whip->audio_par->sample_rate,
- whip->audio_par->ch_layout.nb_channels,
- whip->audio_ssrc,
- whip->audio_ssrc);
- }
-
- if (whip->video_par) {
- level = whip->video_par->level;
- if (whip->video_par->codec_id == AV_CODEC_ID_H264) {
- vcodec_name = "H264";
- profile_iop |= whip->video_par->profile & AV_PROFILE_H264_CONSTRAINED ? 1 << 6 : 0;
- profile_iop |= whip->video_par->profile & AV_PROFILE_H264_INTRA ? 1 << 4 : 0;
- profile_idc = whip->video_par->profile & 0x00ff;
- }
-
- av_bprintf(&bp, ""
- "m=video 9 UDP/TLS/RTP/SAVPF %u %u\r\n"
- "c=IN IP4 0.0.0.0\r\n"
- "a=ice-ufrag:%s\r\n"
- "a=ice-pwd:%s\r\n"
- "a=fingerprint:sha-256 %s\r\n"
- "a=setup:passive\r\n"
- "a=mid:1\r\n"
- "a=sendonly\r\n"
- "a=msid:FFmpeg video\r\n"
- "a=rtcp-mux\r\n"
- "a=rtcp-rsize\r\n"
- "a=rtpmap:%u %s/90000\r\n"
- "a=fmtp:%u level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=%02x%02x%02x\r\n"
- "a=rtcp-fb%u nack\r\n"
- "a=rtpmap:%u rtx/90000\r\n"
- "a=fmtp:%u apt=%u\r\n"
- "a=ssrc-group:FID %u %u\r\n"
- "a=ssrc:%u cname:FFmpeg\r\n"
- "a=ssrc:%u msid:FFmpeg video\r\n",
- whip->video_payload_type,
- whip->video_rtx_payload_type,
- whip->ice_ufrag_local,
- whip->ice_pwd_local,
- whip->dtls_fingerprint,
- whip->video_payload_type,
- vcodec_name,
- whip->video_payload_type,
- profile_idc,
- profile_iop,
- level,
- whip->video_payload_type,
- whip->video_rtx_payload_type,
- whip->video_rtx_payload_type,
- whip->video_payload_type,
- whip->video_ssrc,
- whip->video_rtx_ssrc,
- whip->video_ssrc,
- whip->video_ssrc);
- }
-
- if (!av_bprint_is_complete(&bp)) {
- av_log(whip, AV_LOG_ERROR, "Offer exceed max %d, %s\n", MAX_SDP_SIZE, bp.str);
- ret = AVERROR(EIO);
- goto end;
- }
-
- whip->sdp_offer = av_strdup(bp.str);
- if (!whip->sdp_offer) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
-
- if (whip->state < WHIP_STATE_OFFER)
- whip->state = WHIP_STATE_OFFER;
- whip->whip_offer_time = av_gettime_relative();
- av_log(whip, AV_LOG_VERBOSE, "Generated state=%d, offer: %s\n", whip->state, whip->sdp_offer);
-
-end:
- av_bprint_finalize(&bp, NULL);
- return ret;
-}
-
-/**
- * Exchange SDP offer with WebRTC peer to get the answer.
- *
- * @return 0 if OK, AVERROR_xxx on error
- */
-static int exchange_sdp(AVFormatContext *s)
-{
- int ret;
- char buf[MAX_URL_SIZE];
- AVBPrint bp;
- WHIPContext *whip = s->priv_data;
- /* The URL context is an HTTP transport layer for the WHIP protocol. */
- URLContext *whip_uc = NULL;
- AVDictionary *opts = NULL;
- char *hex_data = NULL;
- const char *proto_name = avio_find_protocol_name(s->url);
-
- /* To prevent a crash during cleanup, always initialize it. */
- av_bprint_init(&bp, 1, MAX_SDP_SIZE);
-
- if (!av_strstart(proto_name, "http", NULL)) {
- av_log(whip, AV_LOG_ERROR, "Protocol %s is not supported by RTC, choose http, url is %s\n",
- proto_name, s->url);
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- if (!whip->sdp_offer || !strlen(whip->sdp_offer)) {
- av_log(whip, AV_LOG_ERROR, "No offer to exchange\n");
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- ret = snprintf(buf, sizeof(buf), "Cache-Control: no-cache\r\nContent-Type: application/sdp\r\n");
- if (whip->authorization)
- ret += snprintf(buf + ret, sizeof(buf) - ret, "Authorization: Bearer %s\r\n", whip->authorization);
- if (ret <= 0 || ret >= sizeof(buf)) {
- av_log(whip, AV_LOG_ERROR, "Failed to generate headers, size=%d, %s\n", ret, buf);
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- av_dict_set(&opts, "headers", buf, 0);
- av_dict_set_int(&opts, "chunked_post", 0, 0);
-
- hex_data = av_mallocz(2 * strlen(whip->sdp_offer) + 1);
- if (!hex_data) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
- ff_data_to_hex(hex_data, whip->sdp_offer, strlen(whip->sdp_offer), 0);
- av_dict_set(&opts, "post_data", hex_data, 0);
-
- ret = ffurl_open_whitelist(&whip_uc, s->url, AVIO_FLAG_READ_WRITE, &s->interrupt_callback,
- &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to request url=%s, offer: %s\n", s->url, whip->sdp_offer);
- goto end;
- }
-
- if (ff_http_get_new_location(whip_uc)) {
- whip->whip_resource_url = av_strdup(ff_http_get_new_location(whip_uc));
- if (!whip->whip_resource_url) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
- }
-
- while (1) {
- ret = ffurl_read(whip_uc, buf, sizeof(buf));
- if (ret == AVERROR_EOF) {
- /* Reset the error because we read all response as answer util EOF. */
- ret = 0;
- break;
- }
- if (ret <= 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to read response from url=%s, offer is %s, answer is %s\n",
- s->url, whip->sdp_offer, whip->sdp_answer);
- goto end;
- }
-
- av_bprintf(&bp, "%.*s", ret, buf);
- if (!av_bprint_is_complete(&bp)) {
- av_log(whip, AV_LOG_ERROR, "Answer exceed max size %d, %.*s, %s\n", MAX_SDP_SIZE, ret, buf, bp.str);
- ret = AVERROR(EIO);
- goto end;
- }
- }
-
- if (!av_strstart(bp.str, "v=", NULL)) {
- av_log(whip, AV_LOG_ERROR, "Invalid answer: %s\n", bp.str);
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- whip->sdp_answer = av_strdup(bp.str);
- if (!whip->sdp_answer) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
-
- if (whip->state < WHIP_STATE_ANSWER)
- whip->state = WHIP_STATE_ANSWER;
- av_log(whip, AV_LOG_VERBOSE, "Got state=%d, answer: %s\n", whip->state, whip->sdp_answer);
-
-end:
- ffurl_closep(&whip_uc);
- av_bprint_finalize(&bp, NULL);
- av_dict_free(&opts);
- av_freep(&hex_data);
- return ret;
-}
-
-/**
- * Parses the ICE ufrag, pwd, and candidates from the SDP answer.
- *
- * This function is used to extract the ICE ufrag, pwd, and candidates from the SDP answer.
- * It returns an error if any of these fields is NULL. The function only uses the first
- * candidate if there are multiple candidates. However, support for multiple candidates
- * will be added in the future.
- *
- * @param s Pointer to the AVFormatContext
- * @returns Returns 0 if successful or AVERROR_xxx if an error occurs.
- */
-static int parse_answer(AVFormatContext *s)
-{
- int ret = 0;
- AVIOContext *pb;
- char line[MAX_URL_SIZE];
- const char *ptr;
- int i;
- WHIPContext *whip = s->priv_data;
-
- if (!whip->sdp_answer || !strlen(whip->sdp_answer)) {
- av_log(whip, AV_LOG_ERROR, "No answer to parse\n");
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- pb = avio_alloc_context(whip->sdp_answer, strlen(whip->sdp_answer), 0, NULL, NULL, NULL, NULL);
- if (!pb)
- return AVERROR(ENOMEM);
-
- for (i = 0; !avio_feof(pb); i++) {
- ff_get_chomp_line(pb, line, sizeof(line));
- if (av_strstart(line, "a=ice-lite", &ptr))
- whip->is_peer_ice_lite = 1;
- if (av_strstart(line, "a=ice-ufrag:", &ptr) && !whip->ice_ufrag_remote) {
- whip->ice_ufrag_remote = av_strdup(ptr);
- if (!whip->ice_ufrag_remote) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
- } else if (av_strstart(line, "a=ice-pwd:", &ptr) && !whip->ice_pwd_remote) {
- whip->ice_pwd_remote = av_strdup(ptr);
- if (!whip->ice_pwd_remote) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
- } else if (av_strstart(line, "a=candidate:", &ptr) && !whip->ice_protocol) {
- if (ptr && av_stristr(ptr, "host")) {
- /* Refer to RFC 5245 15.1 */
- char foundation[33], protocol[17], host[129];
- int component_id, priority, port;
- ret = sscanf(ptr, "%32s %d %16s %d %128s %d typ host", foundation, &component_id, protocol, &priority, host, &port);
- if (ret != 6) {
- av_log(whip, AV_LOG_ERROR, "Failed %d to parse line %d %s from %s\n",
- ret, i, line, whip->sdp_answer);
- ret = AVERROR(EIO);
- goto end;
- }
-
- if (av_strcasecmp(protocol, "udp")) {
- av_log(whip, AV_LOG_ERROR, "Protocol %s is not supported by RTC, choose udp, line %d %s of %s\n",
- protocol, i, line, whip->sdp_answer);
- ret = AVERROR(EIO);
- goto end;
- }
-
- whip->ice_protocol = av_strdup(protocol);
- whip->ice_host = av_strdup(host);
- whip->ice_port = port;
- if (!whip->ice_protocol || !whip->ice_host) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
- }
- }
- }
-
- if (!whip->ice_pwd_remote || !strlen(whip->ice_pwd_remote)) {
- av_log(whip, AV_LOG_ERROR, "No remote ice pwd parsed from %s\n", whip->sdp_answer);
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- if (!whip->ice_ufrag_remote || !strlen(whip->ice_ufrag_remote)) {
- av_log(whip, AV_LOG_ERROR, "No remote ice ufrag parsed from %s\n", whip->sdp_answer);
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- if (!whip->ice_protocol || !whip->ice_host || !whip->ice_port) {
- av_log(whip, AV_LOG_ERROR, "No ice candidate parsed from %s\n", whip->sdp_answer);
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- if (whip->state < WHIP_STATE_NEGOTIATED)
- whip->state = WHIP_STATE_NEGOTIATED;
- whip->whip_answer_time = av_gettime_relative();
- av_log(whip, AV_LOG_VERBOSE, "SDP state=%d, offer=%zuB, answer=%zuB, ufrag=%s, pwd=%zuB, transport=%s://%s:%d, elapsed=%.2fms\n",
- whip->state, strlen(whip->sdp_offer), strlen(whip->sdp_answer), whip->ice_ufrag_remote, strlen(whip->ice_pwd_remote),
- whip->ice_protocol, whip->ice_host, whip->ice_port, ELAPSED(whip->whip_starttime, av_gettime_relative()));
-
-end:
- avio_context_free(&pb);
- return ret;
-}
-
-/**
- * Creates and marshals an ICE binding request packet.
- *
- * This function creates and marshals an ICE binding request packet. The function only
- * generates the username attribute and does not include goog-network-info,
- * use-candidate. However, some of these attributes may be added in the future.
- *
- * @param s Pointer to the AVFormatContext
- * @param buf Pointer to memory buffer to store the request packet
- * @param buf_size Size of the memory buffer
- * @param request_size Pointer to an integer that receives the size of the request packet
- * @return Returns 0 if successful or AVERROR_xxx if an error occurs.
- */
-static int ice_create_request(AVFormatContext *s, uint8_t *buf, int buf_size, int *request_size)
-{
- int ret, size, crc32;
- char username[128];
- AVIOContext *pb = NULL;
- AVHMAC *hmac = NULL;
- WHIPContext *whip = s->priv_data;
-
- pb = avio_alloc_context(buf, buf_size, 1, NULL, NULL, NULL, NULL);
- if (!pb)
- return AVERROR(ENOMEM);
-
- hmac = av_hmac_alloc(AV_HMAC_SHA1);
- if (!hmac) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
-
- /* Write 20 bytes header */
- avio_wb16(pb, 0x0001); /* STUN binding request */
- avio_wb16(pb, 0); /* length */
- avio_wb32(pb, STUN_MAGIC_COOKIE); /* magic cookie */
- avio_wb32(pb, av_lfg_get(&whip->rnd)); /* transaction ID */
- avio_wb32(pb, av_lfg_get(&whip->rnd)); /* transaction ID */
- avio_wb32(pb, av_lfg_get(&whip->rnd)); /* transaction ID */
-
- /* The username is the concatenation of the two ICE ufrag */
- ret = snprintf(username, sizeof(username), "%s:%s", whip->ice_ufrag_remote, whip->ice_ufrag_local);
- if (ret <= 0 || ret >= sizeof(username)) {
- av_log(whip, AV_LOG_ERROR, "Failed to build username %s:%s, max=%zu, ret=%d\n",
- whip->ice_ufrag_remote, whip->ice_ufrag_local, sizeof(username), ret);
- ret = AVERROR(EIO);
- goto end;
- }
-
- /* Write the username attribute */
- avio_wb16(pb, STUN_ATTR_USERNAME); /* attribute type username */
- avio_wb16(pb, ret); /* size of username */
- avio_write(pb, username, ret); /* bytes of username */
- ffio_fill(pb, 0, (4 - (ret % 4)) % 4); /* padding */
-
- /* Write the use-candidate attribute */
- avio_wb16(pb, STUN_ATTR_USE_CANDIDATE); /* attribute type use-candidate */
- avio_wb16(pb, 0); /* size of use-candidate */
-
- avio_wb16(pb, STUN_ATTR_PRIORITY);
- avio_wb16(pb, 4);
- avio_wb32(pb, STUN_HOST_CANDIDATE_PRIORITY);
-
- avio_wb16(pb, STUN_ATTR_ICE_CONTROLLING);
- avio_wb16(pb, 8);
- avio_wb64(pb, whip->ice_tie_breaker);
-
- /* Build and update message integrity */
- avio_wb16(pb, STUN_ATTR_MESSAGE_INTEGRITY); /* attribute type message integrity */
- avio_wb16(pb, 20); /* size of message integrity */
- ffio_fill(pb, 0, 20); /* fill with zero to directly write and skip it */
- size = avio_tell(pb);
- buf[2] = (size - 20) >> 8;
- buf[3] = (size - 20) & 0xFF;
- av_hmac_init(hmac, whip->ice_pwd_remote, strlen(whip->ice_pwd_remote));
- av_hmac_update(hmac, buf, size - 24);
- av_hmac_final(hmac, buf + size - 20, 20);
-
- /* Write the fingerprint attribute */
- avio_wb16(pb, STUN_ATTR_FINGERPRINT); /* attribute type fingerprint */
- avio_wb16(pb, 4); /* size of fingerprint */
- ffio_fill(pb, 0, 4); /* fill with zero to directly write and skip it */
- size = avio_tell(pb);
- buf[2] = (size - 20) >> 8;
- buf[3] = (size - 20) & 0xFF;
- /* Refer to the av_hash_alloc("CRC32"), av_hash_init and av_hash_final */
- crc32 = av_crc(av_crc_get_table(AV_CRC_32_IEEE_LE), 0xFFFFFFFF, buf, size - 8) ^ 0xFFFFFFFF;
- avio_skip(pb, -4);
- avio_wb32(pb, crc32 ^ 0x5354554E); /* xor with "STUN" */
-
- *request_size = size;
-
-end:
- avio_context_free(&pb);
- av_hmac_free(hmac);
- return ret;
-}
-
-/**
- * Create an ICE binding response.
- *
- * This function generates an ICE binding response and writes it to the provided
- * buffer. The response is signed using the local password for message integrity.
- *
- * @param s Pointer to the AVFormatContext structure.
- * @param tid Pointer to the transaction ID of the binding request. The tid_size should be 12.
- * @param tid_size The size of the transaction ID, should be 12.
- * @param buf Pointer to the buffer where the response will be written.
- * @param buf_size The size of the buffer provided for the response.
- * @param response_size Pointer to an integer that will store the size of the generated response.
- * @return Returns 0 if successful or AVERROR_xxx if an error occurs.
- */
-static int ice_create_response(AVFormatContext *s, char *tid, int tid_size, uint8_t *buf, int buf_size, int *response_size)
-{
- int ret = 0, size, crc32;
- AVIOContext *pb = NULL;
- AVHMAC *hmac = NULL;
- WHIPContext *whip = s->priv_data;
-
- if (tid_size != 12) {
- av_log(whip, AV_LOG_ERROR, "Invalid transaction ID size. Expected 12, got %d\n", tid_size);
- return AVERROR(EINVAL);
- }
-
- pb = avio_alloc_context(buf, buf_size, 1, NULL, NULL, NULL, NULL);
- if (!pb)
- return AVERROR(ENOMEM);
-
- hmac = av_hmac_alloc(AV_HMAC_SHA1);
- if (!hmac) {
- ret = AVERROR(ENOMEM);
- goto end;
- }
-
- /* Write 20 bytes header */
- avio_wb16(pb, 0x0101); /* STUN binding response */
- avio_wb16(pb, 0); /* length */
- avio_wb32(pb, STUN_MAGIC_COOKIE); /* magic cookie */
- avio_write(pb, tid, tid_size); /* transaction ID */
-
- /* Build and update message integrity */
- avio_wb16(pb, STUN_ATTR_MESSAGE_INTEGRITY); /* attribute type message integrity */
- avio_wb16(pb, 20); /* size of message integrity */
- ffio_fill(pb, 0, 20); /* fill with zero to directly write and skip it */
- size = avio_tell(pb);
- buf[2] = (size - 20) >> 8;
- buf[3] = (size - 20) & 0xFF;
- av_hmac_init(hmac, whip->ice_pwd_local, strlen(whip->ice_pwd_local));
- av_hmac_update(hmac, buf, size - 24);
- av_hmac_final(hmac, buf + size - 20, 20);
-
- /* Write the fingerprint attribute */
- avio_wb16(pb, STUN_ATTR_FINGERPRINT); /* attribute type fingerprint */
- avio_wb16(pb, 4); /* size of fingerprint */
- ffio_fill(pb, 0, 4); /* fill with zero to directly write and skip it */
- size = avio_tell(pb);
- buf[2] = (size - 20) >> 8;
- buf[3] = (size - 20) & 0xFF;
- /* Refer to the av_hash_alloc("CRC32"), av_hash_init and av_hash_final */
- crc32 = av_crc(av_crc_get_table(AV_CRC_32_IEEE_LE), 0xFFFFFFFF, buf, size - 8) ^ 0xFFFFFFFF;
- avio_skip(pb, -4);
- avio_wb32(pb, crc32 ^ 0x5354554E); /* xor with "STUN" */
-
- *response_size = size;
-
-end:
- avio_context_free(&pb);
- av_hmac_free(hmac);
- return ret;
-}
-
-/**
- * A Binding request has class=0b00 (request) and method=0b000000000001 (Binding)
- * and is encoded into the first 16 bits as 0x0001.
- * See https://datatracker.ietf.org/doc/html/rfc5389#section-6
- */
-static int ice_is_binding_request(uint8_t *b, int size)
-{
- return size >= ICE_STUN_HEADER_SIZE && AV_RB16(&b[0]) == 0x0001;
-}
-
-/**
- * A Binding response has class=0b10 (success response) and method=0b000000000001,
- * and is encoded into the first 16 bits as 0x0101.
- */
-static int ice_is_binding_response(uint8_t *b, int size)
-{
- return size >= ICE_STUN_HEADER_SIZE && AV_RB16(&b[0]) == 0x0101;
-}
-
-/**
- * In RTP packets, the first byte is represented as 0b10xxxxxx, where the initial
- * two bits (0b10) indicate the RTP version,
- * see https://www.rfc-editor.org/rfc/rfc3550#section-5.1
- * The RTCP packet header is similar to RTP,
- * see https://www.rfc-editor.org/rfc/rfc3550#section-6.4.1
- */
-static int media_is_rtp_rtcp(const uint8_t *b, int size)
-{
- return size >= WHIP_RTP_HEADER_SIZE && (b[0] & 0xC0) == 0x80;
-}
-
-/* Whether the packet is RTCP. */
-static int media_is_rtcp(const uint8_t *b, int size)
-{
- return size >= WHIP_RTP_HEADER_SIZE && b[1] >= WHIP_RTCP_PT_START && b[1] <= WHIP_RTCP_PT_END;
-}
-
-/**
- * This function handles incoming binding request messages by responding to them.
- * If the message is not a binding request, it will be ignored.
- */
-static int ice_handle_binding_request(AVFormatContext *s, char *buf, int buf_size)
-{
- int ret = 0, size;
- char tid[12];
- WHIPContext *whip = s->priv_data;
-
- /* Ignore if not a binding request. */
- if (!ice_is_binding_request(buf, buf_size))
- return ret;
-
- if (buf_size < ICE_STUN_HEADER_SIZE) {
- av_log(whip, AV_LOG_ERROR, "Invalid STUN message, expected at least %d, got %d\n",
- ICE_STUN_HEADER_SIZE, buf_size);
- return AVERROR(EINVAL);
- }
-
- /* Parse transaction id from binding request in buf. */
- memcpy(tid, buf + 8, 12);
-
- /* Build the STUN binding response. */
- ret = ice_create_response(s, tid, sizeof(tid), whip->buf, sizeof(whip->buf), &size);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to create STUN binding response, size=%d\n", size);
- return ret;
- }
-
- ret = ffurl_write(whip->udp, whip->buf, size);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to send STUN binding response, size=%d\n", size);
- return ret;
- }
-
- return 0;
-}
-
-/**
- * To establish a connection with the UDP server, we utilize ICE-LITE in a Client-Server
- * mode. In this setup, FFmpeg acts as the UDP client, while the peer functions as the
- * UDP server.
- */
-static int udp_connect(AVFormatContext *s)
-{
- int ret = 0;
- char url[256];
- AVDictionary *opts = NULL;
- WHIPContext *whip = s->priv_data;
-
- /* Build UDP URL and create the UDP context as transport. */
- ff_url_join(url, sizeof(url), "udp", NULL, whip->ice_host, whip->ice_port, NULL);
-
- av_dict_set_int(&opts, "connect", 1, 0);
- av_dict_set_int(&opts, "fifo_size", 0, 0);
- /* Pass through the pkt_size and buffer_size to underling protocol */
- av_dict_set_int(&opts, "pkt_size", whip->pkt_size, 0);
- av_dict_set_int(&opts, "buffer_size", whip->buffer_size, 0);
-
- ret = ffurl_open_whitelist(&whip->udp, url, AVIO_FLAG_WRITE, &s->interrupt_callback,
- &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to connect udp://%s:%d\n", whip->ice_host, whip->ice_port);
- goto end;
- }
-
- /* Make the socket non-blocking, set to READ and WRITE mode after connected */
- ff_socket_nonblock(ffurl_get_file_handle(whip->udp), 1);
- whip->udp->flags |= AVIO_FLAG_READ | AVIO_FLAG_NONBLOCK;
-
- if (whip->state < WHIP_STATE_UDP_CONNECTED)
- whip->state = WHIP_STATE_UDP_CONNECTED;
- whip->whip_udp_time = av_gettime_relative();
- av_log(whip, AV_LOG_VERBOSE, "UDP state=%d, elapsed=%.2fms, connected to udp://%s:%d\n",
- whip->state, ELAPSED(whip->whip_starttime, av_gettime_relative()), whip->ice_host, whip->ice_port);
-
-end:
- av_dict_free(&opts);
- return ret;
-}
-
-static int ice_dtls_handshake(AVFormatContext *s)
-{
- int ret = 0, size, i;
- int64_t starttime = av_gettime_relative(), now;
- WHIPContext *whip = s->priv_data;
- AVDictionary *opts = NULL;
- char buf[256], *cert_buf = NULL, *key_buf = NULL;
-
- if (whip->state < WHIP_STATE_UDP_CONNECTED || !whip->udp) {
- av_log(whip, AV_LOG_ERROR, "UDP not connected, state=%d, udp=%p\n", whip->state, whip->udp);
- return AVERROR(EINVAL);
- }
-
- while (1) {
- if (whip->state <= WHIP_STATE_ICE_CONNECTING) {
- /* Build the STUN binding request. */
- ret = ice_create_request(s, whip->buf, sizeof(whip->buf), &size);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to create STUN binding request, size=%d\n", size);
- goto end;
- }
-
- ret = ffurl_write(whip->udp, whip->buf, size);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to send STUN binding request, size=%d\n", size);
- goto end;
- }
-
- if (whip->state < WHIP_STATE_ICE_CONNECTING)
- whip->state = WHIP_STATE_ICE_CONNECTING;
- }
-
-next_packet:
- if (whip->state >= WHIP_STATE_DTLS_FINISHED)
- /* DTLS handshake is done, exit the loop. */
- break;
-
- now = av_gettime_relative();
- if (now - starttime >= whip->handshake_timeout * WHIP_US_PER_MS) {
- av_log(whip, AV_LOG_ERROR, "DTLS handshake timeout=%dms, cost=%.2fms, elapsed=%.2fms, state=%d\n",
- whip->handshake_timeout, ELAPSED(starttime, now), ELAPSED(whip->whip_starttime, now), whip->state);
- ret = AVERROR(ETIMEDOUT);
- goto end;
- }
-
- /* Read the STUN or DTLS messages from peer. */
- for (i = 0; i < ICE_DTLS_READ_MAX_RETRY; i++) {
- if (whip->state > WHIP_STATE_ICE_CONNECTED)
- break;
- ret = ffurl_read(whip->udp, whip->buf, sizeof(whip->buf));
- if (ret > 0)
- break;
- if (ret == AVERROR(EAGAIN)) {
- av_usleep(ICE_DTLS_READ_SLEEP_DURATION * WHIP_US_PER_MS);
- continue;
- }
- av_log(whip, AV_LOG_ERROR, "Failed to read message\n");
- goto end;
- }
-
- /* Handle the ICE binding response. */
- if (ice_is_binding_response(whip->buf, ret)) {
- if (whip->state < WHIP_STATE_ICE_CONNECTED) {
- if (whip->is_peer_ice_lite)
- whip->state = WHIP_STATE_ICE_CONNECTED;
- whip->whip_ice_time = av_gettime_relative();
- av_log(whip, AV_LOG_VERBOSE, "ICE STUN ok, state=%d, url=udp://%s:%d, location=%s, username=%s:%s, res=%dB, elapsed=%.2fms\n",
- whip->state, whip->ice_host, whip->ice_port, whip->whip_resource_url ? whip->whip_resource_url : "",
- whip->ice_ufrag_remote, whip->ice_ufrag_local, ret, ELAPSED(whip->whip_starttime, av_gettime_relative()));
-
- ff_url_join(buf, sizeof(buf), "dtls", NULL, whip->ice_host, whip->ice_port, NULL);
- av_dict_set_int(&opts, "mtu", whip->pkt_size, 0);
- if (whip->cert_file) {
- av_dict_set(&opts, "cert_file", whip->cert_file, 0);
- } else
- av_dict_set(&opts, "cert_pem", whip->cert_buf, 0);
-
- if (whip->key_file) {
- av_dict_set(&opts, "key_file", whip->key_file, 0);
- } else
- av_dict_set(&opts, "key_pem", whip->key_buf, 0);
- av_dict_set_int(&opts, "external_sock", 1, 0);
- av_dict_set_int(&opts, "use_srtp", 1, 0);
- av_dict_set_int(&opts, "listen", 1, 0);
- /* If got the first binding response, start DTLS handshake. */
- ret = ffurl_open_whitelist(&whip->dtls_uc, buf, AVIO_FLAG_READ_WRITE, &s->interrupt_callback,
- &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
- av_dict_free(&opts);
- if (ret < 0)
- goto end;
- dtls_initialize(s);
- }
- goto next_packet;
- }
-
- /* When a binding request is received, it is necessary to respond immediately. */
- if (ice_is_binding_request(whip->buf, ret)) {
- if ((ret = ice_handle_binding_request(s, whip->buf, ret)) < 0)
- goto end;
- goto next_packet;
- }
-
- /* If got any DTLS messages, handle it. */
- if (is_dtls_packet(whip->buf, ret)) {
- /* Start consent timer when ICE selected */
- whip->whip_last_consent_tx_time = whip->whip_last_consent_rx_time = av_gettime_relative();
- whip->state = WHIP_STATE_ICE_CONNECTED;
- ret = ffurl_handshake(whip->dtls_uc);
- if (ret < 0) {
- whip->state = WHIP_STATE_FAILED;
- av_log(whip, AV_LOG_VERBOSE, "DTLS session failed\n");
- goto end;
- }
- if (!ret) {
- whip->state = WHIP_STATE_DTLS_FINISHED;
- whip->whip_dtls_time = av_gettime_relative();
- av_log(whip, AV_LOG_VERBOSE, "DTLS handshake is done, elapsed=%.2fms\n",
- ELAPSED(whip->whip_starttime, whip->whip_dtls_time));
- }
- goto next_packet;
- }
- }
-
-end:
- if (cert_buf)
- av_free(cert_buf);
- if (key_buf)
- av_free(key_buf);
- return ret;
-}
-
-/**
- * Establish the SRTP context using the keying material exported from DTLS.
- *
- * Create separate SRTP contexts for sending video and audio, as their sequences differ
- * and should not share a single context. Generate a single SRTP context for receiving
- * RTCP only.
- *
- * @return 0 if OK, AVERROR_xxx on error
- */
-static int setup_srtp(AVFormatContext *s)
-{
- int ret;
- char recv_key[DTLS_SRTP_KEY_LEN + DTLS_SRTP_SALT_LEN];
- char send_key[DTLS_SRTP_KEY_LEN + DTLS_SRTP_SALT_LEN];
- char buf[AV_BASE64_SIZE(DTLS_SRTP_KEY_LEN + DTLS_SRTP_SALT_LEN)];
- /**
- * The profile for OpenSSL's SRTP is SRTP_AES128_CM_SHA1_80, see ssl/d1_srtp.c.
- * The profile for FFmpeg's SRTP is SRTP_AES128_CM_HMAC_SHA1_80, see libavformat/srtp.c.
- */
- const char* suite = "SRTP_AES128_CM_HMAC_SHA1_80";
- WHIPContext *whip = s->priv_data;
- ret = ff_dtls_export_materials(whip->dtls_uc, whip->dtls_srtp_materials, sizeof(whip->dtls_srtp_materials));
- if (ret < 0)
- goto end;
- /**
- * This represents the material used to build the SRTP master key. It is
- * generated by DTLS and has the following layout:
- * 16B 16B 14B 14B
- * client_key | server_key | client_salt | server_salt
- */
- char *client_key = whip->dtls_srtp_materials;
- char *server_key = whip->dtls_srtp_materials + DTLS_SRTP_KEY_LEN;
- char *client_salt = server_key + DTLS_SRTP_KEY_LEN;
- char *server_salt = client_salt + DTLS_SRTP_SALT_LEN;
-
- /* As DTLS server, the recv key is client master key plus salt. */
- memcpy(recv_key, client_key, DTLS_SRTP_KEY_LEN);
- memcpy(recv_key + DTLS_SRTP_KEY_LEN, client_salt, DTLS_SRTP_SALT_LEN);
-
- /* As DTLS server, the send key is server master key plus salt. */
- memcpy(send_key, server_key, DTLS_SRTP_KEY_LEN);
- memcpy(send_key + DTLS_SRTP_KEY_LEN, server_salt, DTLS_SRTP_SALT_LEN);
-
- /* Setup SRTP context for outgoing packets */
- if (!av_base64_encode(buf, sizeof(buf), send_key, sizeof(send_key))) {
- av_log(whip, AV_LOG_ERROR, "Failed to encode send key\n");
- ret = AVERROR(EIO);
- goto end;
- }
-
- ret = ff_srtp_set_crypto(&whip->srtp_audio_send, suite, buf);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to set crypto for audio send\n");
- goto end;
- }
-
- ret = ff_srtp_set_crypto(&whip->srtp_video_send, suite, buf);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to set crypto for video send\n");
- goto end;
- }
-
- ret = ff_srtp_set_crypto(&whip->srtp_video_rtx_send, suite, buf);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to set crypto for video rtx send\n");
- goto end;
- }
-
- ret = ff_srtp_set_crypto(&whip->srtp_rtcp_send, suite, buf);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to set crypto for rtcp send\n");
- goto end;
- }
-
- /* Setup SRTP context for incoming packets */
- if (!av_base64_encode(buf, sizeof(buf), recv_key, sizeof(recv_key))) {
- av_log(whip, AV_LOG_ERROR, "Failed to encode recv key\n");
- ret = AVERROR(EIO);
- goto end;
- }
-
- ret = ff_srtp_set_crypto(&whip->srtp_recv, suite, buf);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to set crypto for recv\n");
- goto end;
- }
-
- if (whip->state < WHIP_STATE_SRTP_FINISHED)
- whip->state = WHIP_STATE_SRTP_FINISHED;
- whip->whip_srtp_time = av_gettime_relative();
- av_log(whip, AV_LOG_VERBOSE, "SRTP setup done, state=%d, suite=%s, key=%zuB, elapsed=%.2fms\n",
- whip->state, suite, sizeof(send_key), ELAPSED(whip->whip_starttime, av_gettime_relative()));
-
-end:
- return ret;
-}
/**
* Callback triggered by the RTP muxer when it creates and sends out an RTP packet.
@@ -1641,64 +420,6 @@ end:
return ret;
}
-/**
- * RTC is connectionless, for it's based on UDP, so it check whether sesison is
- * timeout. In such case, publishers can't republish the stream util the session
- * is timeout.
- * This function is called to notify the server that the stream is ended, server
- * should expire and close the session immediately, so that publishers can republish
- * the stream quickly.
- */
-static int dispose_session(AVFormatContext *s)
-{
- int ret;
- char buf[MAX_URL_SIZE];
- URLContext *whip_uc = NULL;
- AVDictionary *opts = NULL;
- WHIPContext *whip = s->priv_data;
-
- if (!whip->whip_resource_url)
- return 0;
-
- ret = snprintf(buf, sizeof(buf), "Cache-Control: no-cache\r\n");
- if (whip->authorization)
- ret += snprintf(buf + ret, sizeof(buf) - ret, "Authorization: Bearer %s\r\n", whip->authorization);
- if (ret <= 0 || ret >= sizeof(buf)) {
- av_log(whip, AV_LOG_ERROR, "Failed to generate headers, size=%d, %s\n", ret, buf);
- ret = AVERROR(EINVAL);
- goto end;
- }
-
- av_dict_set(&opts, "headers", buf, 0);
- av_dict_set_int(&opts, "chunked_post", 0, 0);
- av_dict_set(&opts, "method", "DELETE", 0);
- ret = ffurl_open_whitelist(&whip_uc, whip->whip_resource_url, AVIO_FLAG_READ_WRITE, &s->interrupt_callback,
- &opts, s->protocol_whitelist, s->protocol_blacklist, NULL);
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to DELETE url=%s\n", whip->whip_resource_url);
- goto end;
- }
-
- while (1) {
- ret = ffurl_read(whip_uc, buf, sizeof(buf));
- if (ret == AVERROR_EOF) {
- ret = 0;
- break;
- }
- if (ret < 0) {
- av_log(whip, AV_LOG_ERROR, "Failed to read response from DELETE url=%s\n", whip->whip_resource_url);
- goto end;
- }
- }
-
- av_log(whip, AV_LOG_INFO, "Dispose resource %s ok\n", whip->whip_resource_url);
-
-end:
- ffurl_closep(&whip_uc);
- av_dict_free(&opts);
- return ret;
-}
-
/**
* Since the h264_mp4toannexb filter only processes the MP4 ISOM format and bypasses
* the annexb format, it is necessary to manually insert encoder metadata before each
@@ -1785,28 +506,13 @@ static av_cold int whip_init(AVFormatContext *s)
int ret;
WHIPContext *whip = s->priv_data;
- if ((ret = initialize(s)) < 0)
+ if ((ret = ff_rtc_initialize(s)) < 0)
goto end;
if ((ret = parse_codec(s)) < 0)
goto end;
- if ((ret = generate_sdp_offer(s)) < 0)
- goto end;
-
- if ((ret = exchange_sdp(s)) < 0)
- goto end;
-
- if ((ret = parse_answer(s)) < 0)
- goto end;
-
- if ((ret = udp_connect(s)) < 0)
- goto end;
-
- if ((ret = ice_dtls_handshake(s)) < 0)
- goto end;
-
- if ((ret = setup_srtp(s)) < 0)
+ if ((ret = ff_rtc_connect(s)) < 0)
goto end;
if ((ret = create_rtp_muxer(s)) < 0)
@@ -1861,6 +567,7 @@ static int whip_write_packet(AVFormatContext *s, AVPacket *pkt)
WHIPContext *whip = s->priv_data;
AVStream *st = s->streams[pkt->stream_index];
AVFormatContext *rtp_ctx = st->priv_data;
+
int64_t now = av_gettime_relative();
/**
* Refer to RFC 7675
@@ -1868,7 +575,7 @@ static int whip_write_packet(AVFormatContext *s, AVPacket *pkt)
*/
if (now - whip->whip_last_consent_tx_time > WHIP_ICE_CONSENT_CHECK_INTERVAL * WHIP_US_PER_MS) {
int size;
- ret = ice_create_request(s, whip->buf, sizeof(whip->buf), &size);
+ ret = ff_rtc_ice_create_request(s, whip->buf, sizeof(whip->buf), &size);
if (ret < 0) {
av_log(whip, AV_LOG_ERROR, "Failed to create STUN binding request, size=%d\n", size);
goto end;
@@ -1897,11 +604,13 @@ static int whip_write_packet(AVFormatContext *s, AVPacket *pkt)
av_log(whip, AV_LOG_ERROR, "Receive EOF from UDP socket\n");
goto end;
}
- if (ice_is_binding_response(whip->buf, ret)) {
+
+ if (ff_rtc_ice_is_binding_response(whip->buf, ret)) {
whip->whip_last_consent_rx_time = av_gettime_relative();
av_log(whip, AV_LOG_DEBUG, "Consent Freshness check received\n");
}
- if (is_dtls_packet(whip->buf, ret)) {
+
+ if (ff_rtc_is_dtls_packet(whip->buf, ret)) {
if ((ret = ffurl_write(whip->dtls_uc, whip->buf, ret)) < 0) {
av_log(whip, AV_LOG_ERROR, "Failed to handle DTLS message\n");
goto end;
@@ -1929,6 +638,7 @@ write_packet:
ret = AVERROR(ETIMEDOUT);
goto end;
}
+
if (whip->h264_annexb_insert_sps_pps && st->codecpar->codec_id == AV_CODEC_ID_H264) {
if ((ret = h264_annexb_insert_sps_pps(s, pkt)) < 0) {
av_log(whip, AV_LOG_ERROR, "Failed to insert SPS/PPS before IDR\n");
@@ -1956,47 +666,7 @@ end:
static av_cold void whip_deinit(AVFormatContext *s)
{
- int i, ret;
- WHIPContext *whip = s->priv_data;
-
- ret = dispose_session(s);
- if (ret < 0)
- av_log(whip, AV_LOG_WARNING, "Failed to dispose resource, ret=%d\n", ret);
-
- for (i = 0; i < s->nb_streams; i++) {
- AVFormatContext* rtp_ctx = s->streams[i]->priv_data;
- if (!rtp_ctx)
- continue;
-
- av_write_trailer(rtp_ctx);
- /**
- * Keep in mind that it is necessary to free the buffer of pb since we allocate
- * it and pass it to pb using avio_alloc_context, while avio_context_free does
- * not perform this action.
- */
- av_freep(&rtp_ctx->pb->buffer);
- avio_context_free(&rtp_ctx->pb);
- avformat_free_context(rtp_ctx);
- s->streams[i]->priv_data = NULL;
- }
-
- av_freep(&whip->sdp_offer);
- av_freep(&whip->sdp_answer);
- av_freep(&whip->whip_resource_url);
- av_freep(&whip->ice_ufrag_remote);
- av_freep(&whip->ice_pwd_remote);
- av_freep(&whip->ice_protocol);
- av_freep(&whip->ice_host);
- av_freep(&whip->authorization);
- av_freep(&whip->cert_file);
- av_freep(&whip->key_file);
- ff_srtp_free(&whip->srtp_audio_send);
- ff_srtp_free(&whip->srtp_video_send);
- ff_srtp_free(&whip->srtp_video_rtx_send);
- ff_srtp_free(&whip->srtp_rtcp_send);
- ff_srtp_free(&whip->srtp_recv);
- ffurl_close(whip->dtls_uc);
- ffurl_closep(&whip->udp);
+ ff_rtc_close(s);
}
static int whip_check_bitstream(AVFormatContext *s, AVStream *st, const AVPacket *pkt)
@@ -2018,22 +688,10 @@ static int whip_check_bitstream(AVFormatContext *s, AVStream *st, const AVPacket
return ret;
}
-#define OFFSET(x) offsetof(WHIPContext, x)
-#define ENC AV_OPT_FLAG_ENCODING_PARAM
-static const AVOption options[] = {
- { "handshake_timeout", "Timeout in milliseconds for ICE and DTLS handshake.", OFFSET(handshake_timeout), AV_OPT_TYPE_INT, { .i64 = 5000 }, -1, INT_MAX, ENC },
- { "pkt_size", "The maximum size, in bytes, of RTP packets that send out", OFFSET(pkt_size), AV_OPT_TYPE_INT, { .i64 = 1200 }, -1, INT_MAX, ENC },
- { "buffer_size", "The buffer size, in bytes, of underlying protocol", OFFSET(buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, ENC },
- { "authorization", "The optional Bearer token for WHIP Authorization", OFFSET(authorization), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, ENC },
- { "cert_file", "The optional certificate file path for DTLS", OFFSET(cert_file), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, ENC },
- { "key_file", "The optional private key file path for DTLS", OFFSET(key_file), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, ENC },
- { NULL },
-};
-
static const AVClass whip_muxer_class = {
.class_name = "WHIP muxer",
.item_name = av_default_item_name,
- .option = options,
+ .option = ff_rtc_options,
.version = LIBAVUTIL_VERSION_INT,
};
--
2.51.0
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next parent reply other threads:[~2025-10-12 15:42 UTC|newest]
Thread overview: 3+ messages / expand[flat|nested] mbox.gz Atom feed top
[not found] <20251012152347.1022477-1-1007668733@qq.com>
2025-10-12 15:41 ` baigao via ffmpeg-devel [this message]
2025-10-12 15:42 ` [FFmpeg-devel] [PATCH 2/3] avformat/whip whep: reanme whip prefix to rtc for common RTC structures baigao via ffmpeg-devel
2025-10-12 15:42 ` [FFmpeg-devel] [PATCH 3/3] avformat/whip whep: add whep support baigao via ffmpeg-devel
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