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no-senders; approved; loop; banned-address; header-match-ffmpeg-devel.ffmpeg.org-0; header-match-ffmpeg-devel.ffmpeg.org-1; header-match-ffmpeg-devel.ffmpeg.org-2; header-match-ffmpeg-devel.ffmpeg.org-3; emergency; member-moderation; nonmember-moderation; administrivia; implicit-dest; max-recipients; max-size; news-moderation; no-subject; digests; suspicious-header X-Mailman-Version: 3.3.10 Precedence: list Reply-To: FFmpeg development discussions and patches Subject: [FFmpeg-devel] [PATCH 1/3] avformat/whip whep: create rtc for common RTC code shared by whip and whep List-Id: FFmpeg development discussions and patches Archived-At: Archived-At: List-Archive: List-Archive: List-Help: List-Owner: List-Post: List-Subscribe: List-Unsubscribe: From: baigao via ffmpeg-devel Cc: baigao <1007668733@qq.com> Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Archived-At: List-Archive: List-Post: --- 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 +#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 _______________________________________________ ffmpeg-devel mailing list -- ffmpeg-devel@ffmpeg.org To unsubscribe send an email to ffmpeg-devel-leave@ffmpeg.org