## Introduce This PR adds support for viewing streams via the RTSP protocol. Note that it only supports viewing streams, not publishing streams via RTSP. Currently, only publishing via RTMP is supported, which is then converted to RTSP. Further work is needed to support publishing RTC/SRT streams and converting them to RTSP. ## Usage Build and run SRS with RTSP support: ``` cd srs/trunk && ./configure --rtsp=on && make -j16 ./objs/srs -c conf/rtsp.conf ``` Push stream via RTMP by FFmpeg: ``` ffmpeg -re -i doc/source.flv -c copy -f flv rtmp://localhost/live/livestream ``` View the stream via RTSP protocol, try UDP first, then use TCP: ``` ffplay -i rtsp://localhost:8554/live/livestream ``` Or specify the transport protocol with TCP: ``` ffplay -rtsp_transport tcp -i rtsp://localhost:8554/live/livestream ``` ## Unit Test Run utest for RTSP: ``` ./configure --utest=on & make utest -j16 ./objs/srs_utest ``` ## Regression Test You need to start SRS for regression testing. ``` ./objs/srs -c conf/regression-test-for-clion.conf ``` Then run regression tests for RTSP. ``` cd srs/trunk/3rdparty/srs-bench go test ./srs -mod=vendor -v -count=1 -run=TestRtmpPublish_RtspPlay ``` ## Blackbox Test For blackbox testing, SRS will be started by utest, so there is no need to start SRS manually. ``` cd srs/trunk/3rdparty/srs-bench go test ./blackbox -mod=vendor -v -count=1 -run=TestFast_RtmpPublish_RtspPlay_Basic ``` ## UDP Transport As UDP requires port allocation, this PR doesn't support delivering media stream via UDP transport, so it will fail if you try to use UDP as transport: ``` ffplay -rtsp_transport udp -i rtsp://localhost:8554/live/livestream [rtsp @ 0x7fbc99a14880] method SETUP failed: 461 Unsupported Transport rtsp://localhost:8554/live/livestream: Protocol not supported [2025-07-05 21:30:52.738][WARN][14916][7d7gf623][35] RTSP: setup failed: code=2057 (RtspTransportNotSupported) : UDP transport not supported, only TCP/interleaved mode is supported ``` There are no plans to support UDP transport for RTSP. In the real world, UDP is rarely used; the vast majority of RTSP traffic uses TCP. ## Play Before Publish RTSP supports audio with AAC and OPUS codecs, which is significantly different from RTMP or WebRTC. RTSP uses commands to exchange SDP and specify the audio track to play, unlike WHEP or HTTP-FLV, which use the query string of the URL. RTSP depends on the player’s behavior, making it very difficult to use and describe. Considering the feature that allows playing the stream before publishing it, it requires generating some default parameters in the SDP. For OPUS, the sample rate is 48 kHz with 2 channels, while AAC is more complex, especially regarding the sample rate, which may be 44.1 kHz, 32 kHz, or 48 kHz. Therefore, for RTSP, we cannot support play-then-publish. Instead, there must already be a stream when playing it, so that the audio codec is determined. ## Opus Codec No Opus codec support for RTSP, because for RTC2RTSP, it always converts RTC to RTMP frames, then converts them to RTSP packets. Therefore, the audio codec is always AAC after converting RTC to RTMP. This means the bridge architecture needs some changes. We need a new bridge that binds to the target protocol. For example, RTC2RTMP converts the audio codec, but RTC2RTSP keeps the original audio codec. Furthermore, the RTC2RTMP bridge should also support bypassing the Opus codec if we use enhanced-RTMP, which supports the Opus audio codec. I think it should be configurable to either transcode or bypass the audio codec. However, this is not relevant to RTSP. ## AI Contributor Below commits are contributed by AI: * [AI: Remove support for media transport via UDP.](755686229f) * [AI: Add crutial logs for each RTSP stage.](9c8cbe7bde) * [AI: Support AAC doec for RTSP.](7d7cc12bae) * [AI: Add option --rtsp for RTSP.](f67414d9ee) * [AI: Extract SrsRtpVideoBuilder for RTC and RTSP.](562e76b904) --------- Co-authored-by: Jacob Su <suzp1984@gmail.com> Co-authored-by: winlin <winlinvip@gmail.com>
751 lines
25 KiB
C++
751 lines
25 KiB
C++
//
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// Copyright (c) 2013-2025 The SRS Authors
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//
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// SPDX-License-Identifier: MIT
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//
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#include <srs_protocol_rtsp_stack.hpp>
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#include <srs_protocol_io.hpp>
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#include <srs_kernel_stream.hpp>
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#include <srs_kernel_error.hpp>
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#include <srs_kernel_log.hpp>
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#include <srs_kernel_consts.hpp>
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#include <srs_core_autofree.hpp>
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#include <srs_kernel_utility.hpp>
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#include <srs_kernel_buffer.hpp>
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#include <srs_kernel_codec.hpp>
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#include <stdlib.h>
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#include <map>
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using namespace std;
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#define SRS_RTSP_BUFFER 4096
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// Forward declaration of RTCP detection function
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extern bool srs_is_rtcp(const uint8_t* data, size_t len);
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// get the status text of code.
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string srs_generate_rtsp_status_text(int status)
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{
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static std::map<int, std::string> _status_map;
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if (_status_map.empty()) {
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_status_map[SRS_CONSTS_RTSP_Continue] = SRS_CONSTS_RTSP_Continue_str;
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_status_map[SRS_CONSTS_RTSP_OK] = SRS_CONSTS_RTSP_OK_str;
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_status_map[SRS_CONSTS_RTSP_Created] = SRS_CONSTS_RTSP_Created_str;
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_status_map[SRS_CONSTS_RTSP_LowOnStorageSpace] = SRS_CONSTS_RTSP_LowOnStorageSpace_str;
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_status_map[SRS_CONSTS_RTSP_MultipleChoices] = SRS_CONSTS_RTSP_MultipleChoices_str;
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_status_map[SRS_CONSTS_RTSP_MovedPermanently] = SRS_CONSTS_RTSP_MovedPermanently_str;
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_status_map[SRS_CONSTS_RTSP_MovedTemporarily] = SRS_CONSTS_RTSP_MovedTemporarily_str;
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_status_map[SRS_CONSTS_RTSP_SeeOther] = SRS_CONSTS_RTSP_SeeOther_str;
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_status_map[SRS_CONSTS_RTSP_NotModified] = SRS_CONSTS_RTSP_NotModified_str;
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_status_map[SRS_CONSTS_RTSP_UseProxy] = SRS_CONSTS_RTSP_UseProxy_str;
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_status_map[SRS_CONSTS_RTSP_BadRequest] = SRS_CONSTS_RTSP_BadRequest_str;
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_status_map[SRS_CONSTS_RTSP_Unauthorized] = SRS_CONSTS_RTSP_Unauthorized_str;
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_status_map[SRS_CONSTS_RTSP_PaymentRequired] = SRS_CONSTS_RTSP_PaymentRequired_str;
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_status_map[SRS_CONSTS_RTSP_Forbidden] = SRS_CONSTS_RTSP_Forbidden_str;
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_status_map[SRS_CONSTS_RTSP_NotFound] = SRS_CONSTS_RTSP_NotFound_str;
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_status_map[SRS_CONSTS_RTSP_MethodNotAllowed] = SRS_CONSTS_RTSP_MethodNotAllowed_str;
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_status_map[SRS_CONSTS_RTSP_NotAcceptable] = SRS_CONSTS_RTSP_NotAcceptable_str;
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_status_map[SRS_CONSTS_RTSP_ProxyAuthenticationRequired] = SRS_CONSTS_RTSP_ProxyAuthenticationRequired_str;
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_status_map[SRS_CONSTS_RTSP_RequestTimeout] = SRS_CONSTS_RTSP_RequestTimeout_str;
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_status_map[SRS_CONSTS_RTSP_Gone] = SRS_CONSTS_RTSP_Gone_str;
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_status_map[SRS_CONSTS_RTSP_LengthRequired] = SRS_CONSTS_RTSP_LengthRequired_str;
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_status_map[SRS_CONSTS_RTSP_PreconditionFailed] = SRS_CONSTS_RTSP_PreconditionFailed_str;
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_status_map[SRS_CONSTS_RTSP_RequestEntityTooLarge] = SRS_CONSTS_RTSP_RequestEntityTooLarge_str;
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_status_map[SRS_CONSTS_RTSP_RequestURITooLarge] = SRS_CONSTS_RTSP_RequestURITooLarge_str;
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_status_map[SRS_CONSTS_RTSP_UnsupportedMediaType] = SRS_CONSTS_RTSP_UnsupportedMediaType_str;
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_status_map[SRS_CONSTS_RTSP_ParameterNotUnderstood] = SRS_CONSTS_RTSP_ParameterNotUnderstood_str;
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_status_map[SRS_CONSTS_RTSP_ConferenceNotFound] = SRS_CONSTS_RTSP_ConferenceNotFound_str;
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_status_map[SRS_CONSTS_RTSP_NotEnoughBandwidth] = SRS_CONSTS_RTSP_NotEnoughBandwidth_str;
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_status_map[SRS_CONSTS_RTSP_SessionNotFound] = SRS_CONSTS_RTSP_SessionNotFound_str;
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_status_map[SRS_CONSTS_RTSP_MethodNotValidInThisState] = SRS_CONSTS_RTSP_MethodNotValidInThisState_str;
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_status_map[SRS_CONSTS_RTSP_HeaderFieldNotValidForResource] = SRS_CONSTS_RTSP_HeaderFieldNotValidForResource_str;
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_status_map[SRS_CONSTS_RTSP_InvalidRange] = SRS_CONSTS_RTSP_InvalidRange_str;
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_status_map[SRS_CONSTS_RTSP_ParameterIsReadOnly] = SRS_CONSTS_RTSP_ParameterIsReadOnly_str;
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_status_map[SRS_CONSTS_RTSP_AggregateOperationNotAllowed] = SRS_CONSTS_RTSP_AggregateOperationNotAllowed_str;
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_status_map[SRS_CONSTS_RTSP_OnlyAggregateOperationAllowed] = SRS_CONSTS_RTSP_OnlyAggregateOperationAllowed_str;
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_status_map[SRS_CONSTS_RTSP_UnsupportedTransport] = SRS_CONSTS_RTSP_UnsupportedTransport_str;
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_status_map[SRS_CONSTS_RTSP_DestinationUnreachable] = SRS_CONSTS_RTSP_DestinationUnreachable_str;
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_status_map[SRS_CONSTS_RTSP_InternalServerError] = SRS_CONSTS_RTSP_InternalServerError_str;
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_status_map[SRS_CONSTS_RTSP_NotImplemented] = SRS_CONSTS_RTSP_NotImplemented_str;
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_status_map[SRS_CONSTS_RTSP_BadGateway] = SRS_CONSTS_RTSP_BadGateway_str;
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_status_map[SRS_CONSTS_RTSP_ServiceUnavailable] = SRS_CONSTS_RTSP_ServiceUnavailable_str;
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_status_map[SRS_CONSTS_RTSP_GatewayTimeout] = SRS_CONSTS_RTSP_GatewayTimeout_str;
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_status_map[SRS_CONSTS_RTSP_RTSPVersionNotSupported] = SRS_CONSTS_RTSP_RTSPVersionNotSupported_str;
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_status_map[SRS_CONSTS_RTSP_OptionNotSupported] = SRS_CONSTS_RTSP_OptionNotSupported_str;
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}
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std::string status_text;
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if (_status_map.find(status) == _status_map.end()) {
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status_text = "Status Unknown";
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} else {
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status_text = _status_map[status];
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}
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return status_text;
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}
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std::string srs_generate_rtsp_method_str(SrsRtspMethod method)
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{
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switch (method) {
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case SrsRtspMethodDescribe: return SRS_RTSP_METHOD_DESCRIBE;
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case SrsRtspMethodAnnounce: return SRS_RTSP_METHOD_ANNOUNCE;
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case SrsRtspMethodGetParameter: return SRS_RTSP_METHOD_GET_PARAMETER;
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case SrsRtspMethodOptions: return SRS_RTSP_METHOD_OPTIONS;
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case SrsRtspMethodPause: return SRS_RTSP_METHOD_PAUSE;
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case SrsRtspMethodPlay: return SRS_RTSP_METHOD_PLAY;
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case SrsRtspMethodRecord: return SRS_RTSP_METHOD_RECORD;
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case SrsRtspMethodRedirect: return SRS_RTSP_METHOD_REDIRECT;
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case SrsRtspMethodSetup: return SRS_RTSP_METHOD_SETUP;
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case SrsRtspMethodSetParameter: return SRS_RTSP_METHOD_SET_PARAMETER;
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case SrsRtspMethodTeardown: return SRS_RTSP_METHOD_TEARDOWN;
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default: return "Unknown";
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}
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}
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SrsRtspTransport::SrsRtspTransport()
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{
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client_port_min = 0;
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client_port_max = 0;
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interleaved_min = 0;
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interleaved_max = 0;
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}
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SrsRtspTransport::~SrsRtspTransport()
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{
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}
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srs_error_t SrsRtspTransport::parse(string attr)
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{
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srs_error_t err = srs_success;
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size_t pos = string::npos;
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std::string token = attr;
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while (!token.empty()) {
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std::string item = token;
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if ((pos = item.find(";")) != string::npos) {
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item = token.substr(0, pos);
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token = token.substr(pos + 1);
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} else {
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token = "";
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}
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std::string item_key = item, item_value;
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if ((pos = item.find("=")) != string::npos) {
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item_key = item.substr(0, pos);
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item_value = item.substr(pos + 1);
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}
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if (transport.empty() && item.find("=") == string::npos && item_key != "unicast" && item_key != "multicast") {
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transport = item_key;
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if ((pos = transport.find("/")) != string::npos) {
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profile = transport.substr(pos + 1);
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transport = transport.substr(0, pos);
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}
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if ((pos = profile.find("/")) != string::npos) {
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lower_transport = profile.substr(pos + 1);
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profile = profile.substr(0, pos);
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}
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}
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if (item_key == "unicast" || item_key == "multicast") {
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cast_type = item_key;
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} else if (item_key == "interleaved") {
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interleaved = item_value;
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if ((pos = interleaved.find("-")) != string::npos) {
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interleaved_min = ::atoi(interleaved.substr(0, pos).c_str());
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interleaved_max = ::atoi(interleaved.substr(pos + 1).c_str());
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}
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} else if (item_key == "mode") {
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mode = item_value;
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} else if (item_key == "client_port") {
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std::string sport = item_value;
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std::string eport = item_value;
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if ((pos = eport.find("-")) != string::npos) {
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sport = eport.substr(0, pos);
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eport = eport.substr(pos + 1);
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}
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client_port_min = ::atoi(sport.c_str());
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client_port_max = ::atoi(eport.c_str());
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}
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}
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return err;
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}
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void SrsRtspTransport::copy(SrsRtspTransport *src)
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{
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transport = src->transport;
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profile = src->profile;
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lower_transport = src->lower_transport;
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cast_type = src->cast_type;
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interleaved = src->interleaved;
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mode = src->mode;
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}
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SrsRtspRequest::SrsRtspRequest()
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{
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seq = 0;
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content_length = 0;
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stream_id = 0;
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transport = NULL;
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}
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SrsRtspRequest::~SrsRtspRequest()
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{
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srs_freep(transport);
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}
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bool SrsRtspRequest::is_options()
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{
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return method == SRS_RTSP_METHOD_OPTIONS;
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}
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bool SrsRtspRequest::is_describe()
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{
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return method == SRS_RTSP_METHOD_DESCRIBE;
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}
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bool SrsRtspRequest::is_setup()
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{
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return method == SRS_RTSP_METHOD_SETUP;
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}
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bool SrsRtspRequest::is_play()
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{
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return method == SRS_RTSP_METHOD_PLAY;
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}
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bool SrsRtspRequest::is_teardown()
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{
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return method == SRS_RTSP_METHOD_TEARDOWN;
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}
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SrsRtspResponse::SrsRtspResponse(int cseq)
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{
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seq = cseq;
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status = SRS_CONSTS_RTSP_OK;
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}
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SrsRtspResponse::~SrsRtspResponse()
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{
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}
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srs_error_t SrsRtspResponse::encode(stringstream& ss)
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{
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srs_error_t err = srs_success;
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// status line
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ss << SRS_RTSP_VERSION << SRS_RTSP_SP
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<< status << SRS_RTSP_SP
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<< srs_generate_rtsp_status_text(status) << SRS_RTSP_CRLF;
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// cseq
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ss << SRS_RTSP_TOKEN_CSEQ << ":" << SRS_RTSP_SP << seq << SRS_RTSP_CRLF;
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// others.
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ss << "Cache-Control: no-store" << SRS_RTSP_CRLF
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<< "Pragma: no-cache" << SRS_RTSP_CRLF
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<< "Server: " << RTMP_SIG_SRS_SERVER << SRS_RTSP_CRLF;
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// session if specified.
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if (!session.empty()) {
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ss << SRS_RTSP_TOKEN_SESSION << ":" << SRS_RTSP_SP << session << SRS_RTSP_CRLF;
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}
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if ((err = encode_header(ss)) != srs_success) {
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return srs_error_wrap(err, "encode header");
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};
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// header EOF.
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ss << SRS_RTSP_CRLF;
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return err;
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}
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srs_error_t SrsRtspResponse::encode_header(std::stringstream& ss)
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{
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return srs_success;
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}
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SrsRtspOptionsResponse::SrsRtspOptionsResponse(int cseq) : SrsRtspResponse(cseq)
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{
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methods = (SrsRtspMethod)(SrsRtspMethodDescribe | SrsRtspMethodOptions
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| SrsRtspMethodPlay | SrsRtspMethodSetup | SrsRtspMethodTeardown);
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}
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SrsRtspOptionsResponse::~SrsRtspOptionsResponse()
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{
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}
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srs_error_t SrsRtspOptionsResponse::encode_header(stringstream& ss)
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{
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static const SrsRtspMethod rtsp_methods[] = {
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SrsRtspMethodDescribe,
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SrsRtspMethodGetParameter,
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SrsRtspMethodOptions,
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SrsRtspMethodPause,
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SrsRtspMethodPlay,
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SrsRtspMethodRedirect,
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SrsRtspMethodSetup,
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SrsRtspMethodSetParameter,
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SrsRtspMethodTeardown,
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};
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ss << SRS_RTSP_TOKEN_PUBLIC << ":" << SRS_RTSP_SP;
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bool appended = false;
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int nb_methods = (int)(sizeof(rtsp_methods) / sizeof(SrsRtspMethod));
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for (int i = 0; i < nb_methods; i++) {
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SrsRtspMethod method = rtsp_methods[i];
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if (((int)methods & (int)method) == 0) {
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continue;
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}
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if (appended) {
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ss << ", ";
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}
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ss << srs_generate_rtsp_method_str(method);
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appended = true;
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}
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ss << SRS_RTSP_CRLF;
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return srs_success;
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}
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SrsRtspDescribeResponse::SrsRtspDescribeResponse(int cseq) : SrsRtspResponse(cseq)
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{
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}
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SrsRtspDescribeResponse::~SrsRtspDescribeResponse()
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{
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}
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srs_error_t SrsRtspDescribeResponse::encode_header(stringstream& ss)
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{
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ss << SRS_RTSP_TOKEN_CONTENT_TYPE << ":" << SRS_RTSP_SP << "application/sdp" << SRS_RTSP_CRLF;
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// WILL add CRLF to the end of sdp in SrsRtspResponse::encode, so add 2.
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ss << SRS_RTSP_TOKEN_CONTENT_LENGTH << ":" << SRS_RTSP_SP << sdp.length() + 2 << SRS_RTSP_CRLF;
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ss << SRS_RTSP_CRLF;
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ss << sdp;
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return srs_success;
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}
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SrsRtspSetupResponse::SrsRtspSetupResponse(int seq) : SrsRtspResponse(seq)
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{
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transport = new SrsRtspTransport();
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local_port_min = 0;
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local_port_max = 0;
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client_port_min = 0;
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client_port_max = 0;
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}
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SrsRtspSetupResponse::~SrsRtspSetupResponse()
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{
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srs_freep(transport);
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}
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srs_error_t SrsRtspSetupResponse::encode_header(stringstream& ss)
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{
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ss << SRS_RTSP_TOKEN_TRANSPORT << ":" << SRS_RTSP_SP;
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ss << transport->transport << "/" << transport->profile;
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if (!transport->lower_transport.empty()) {
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ss << "/" << transport->lower_transport;
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}
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if (!transport->cast_type.empty()) {
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ss << ";" << transport->cast_type;
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}
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if (!transport->interleaved.empty()) {
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ss << ";interleaved=" << transport->interleaved;
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}
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|
|
if (transport->lower_transport != "TCP") {
|
|
ss << ";client_port=" << client_port_min << "-" << client_port_max;
|
|
ss << ";server_port=" << local_port_min << "-" << local_port_max;
|
|
}
|
|
|
|
ss << ";ssrc=" << ssrc << ";mode=\"play\"";
|
|
|
|
ss << SRS_RTSP_CRLF;
|
|
|
|
return srs_success;
|
|
}
|
|
|
|
SrsRtspPlayResponse::SrsRtspPlayResponse(int cseq) : SrsRtspResponse(cseq)
|
|
{
|
|
}
|
|
|
|
SrsRtspPlayResponse::~SrsRtspPlayResponse()
|
|
{
|
|
}
|
|
|
|
srs_error_t SrsRtspPlayResponse::encode_header(stringstream& ss)
|
|
{
|
|
return srs_success;
|
|
}
|
|
|
|
SrsRtspStack::SrsRtspStack(ISrsProtocolReadWriter* s)
|
|
{
|
|
buf = new SrsSimpleStream();
|
|
skt = s;
|
|
}
|
|
|
|
SrsRtspStack::~SrsRtspStack()
|
|
{
|
|
srs_freep(buf);
|
|
}
|
|
|
|
srs_error_t SrsRtspStack::recv_message(SrsRtspRequest** preq)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRtspRequest* req = new SrsRtspRequest();
|
|
if ((err = do_recv_message(req)) != srs_success) {
|
|
srs_freep(req);
|
|
return srs_error_wrap(err, "recv message");
|
|
}
|
|
|
|
*preq = req;
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtspStack::send_message(SrsRtspResponse* res)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
std::stringstream ss;
|
|
// encode the message to string.
|
|
if ((err = res->encode(ss)) != srs_success) {
|
|
return srs_error_wrap(err, "encode message");
|
|
}
|
|
|
|
std::string str = ss.str();
|
|
srs_assert(!str.empty());
|
|
|
|
if ((err = skt->write((char*)str.c_str(), (int)str.length(), NULL)) != srs_success) {
|
|
return srs_error_wrap(err, "write message");
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtspStack::do_recv_message(SrsRtspRequest* req)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// Parse RTSP request line: "METHOD URI VERSION"
|
|
// Example: "PLAY rtsp://example.com/stream RTSP/1.0"
|
|
|
|
// Parse the RTSP method (PLAY, SETUP, DESCRIBE, etc.)
|
|
if ((err = recv_token_normal(req->method)) != srs_success) {
|
|
return srs_error_wrap(err, "method");
|
|
}
|
|
|
|
// Parse the request URI (resource path or full URL)
|
|
if ((err = recv_token_normal(req->uri)) != srs_success) {
|
|
return srs_error_wrap(err, "uri");
|
|
}
|
|
|
|
// Parse the RTSP version (typically "RTSP/1.0")
|
|
if ((err = recv_token_eof(req->version)) != srs_success) {
|
|
return srs_error_wrap(err, "version");
|
|
}
|
|
|
|
// Parse RTSP headers in "Name: Value" format
|
|
// Example headers:
|
|
// CSeq: 1
|
|
// Content-Type: application/sdp
|
|
// Content-Length: 460
|
|
// Transport: RTP/AVP;unicast;client_port=8000-8001
|
|
// Session: 12345678
|
|
for (;;) {
|
|
// Parse the header name (before the colon)
|
|
std::string token;
|
|
if ((err = recv_token_normal(token)) != srs_success) {
|
|
if (srs_error_code(err) == ERROR_RTSP_REQUEST_HEADER_EOF) {
|
|
srs_error_reset(err);
|
|
break; // End of headers reached (empty line)
|
|
}
|
|
return srs_error_wrap(err, "recv token");
|
|
}
|
|
|
|
// Parse the header value (after the colon) based on header name
|
|
if (token == SRS_RTSP_TOKEN_CSEQ) {
|
|
// CSeq: sequence number for request/response matching
|
|
std::string seq;
|
|
if ((err = recv_token_eof(seq)) != srs_success) {
|
|
return srs_error_wrap(err, "seq");
|
|
}
|
|
req->seq = ::atoll(seq.c_str());
|
|
} else if (token == SRS_RTSP_TOKEN_CONTENT_TYPE) {
|
|
// Content-Type: MIME type of the message body (e.g., application/sdp)
|
|
std::string ct;
|
|
if ((err = recv_token_eof(ct)) != srs_success) {
|
|
return srs_error_wrap(err, "ct");
|
|
}
|
|
req->content_type = ct;
|
|
} else if (token == SRS_RTSP_TOKEN_CONTENT_LENGTH) {
|
|
// Content-Length: size of the message body in bytes
|
|
std::string cl;
|
|
if ((err = recv_token_eof(cl)) != srs_success) {
|
|
return srs_error_wrap(err, "cl");
|
|
}
|
|
req->content_length = ::atoll(cl.c_str());
|
|
} else if (token == SRS_RTSP_TOKEN_TRANSPORT) {
|
|
// Transport: RTP transport parameters (protocol, ports, etc.)
|
|
std::string transport;
|
|
if ((err = recv_token_eof(transport)) != srs_success) {
|
|
return srs_error_wrap(err, "transport");
|
|
}
|
|
if (!req->transport) {
|
|
req->transport = new SrsRtspTransport();
|
|
}
|
|
if ((err = req->transport->parse(transport)) != srs_success) {
|
|
return srs_error_wrap(err, "parse transport=%s", transport.c_str());
|
|
}
|
|
} else if (token == SRS_RTSP_TOKEN_SESSION) {
|
|
// Session: session identifier for maintaining state
|
|
if ((err = recv_token_eof(req->session)) != srs_success) {
|
|
return srs_error_wrap(err, "session");
|
|
}
|
|
} else if (token == SRS_RTSP_TOKEN_ACCEPT) {
|
|
// Accept: acceptable media types for the response
|
|
if ((err = recv_token_eof(req->accept)) != srs_success) {
|
|
return srs_error_wrap(err, "accept");
|
|
}
|
|
} else if (token == SRS_RTSP_TOKEN_USER_AGENT) {
|
|
// User-Agent: client software identification
|
|
if ((err = recv_token_util_eof(req->user_agent)) != srs_success) {
|
|
return srs_error_wrap(err, "user_agent");
|
|
}
|
|
} else if (token == SRS_RTSP_TOKEN_RANGE) {
|
|
// Range: time range for playback (e.g., npt=0-30)
|
|
if ((err = recv_token_eof(req->range)) != srs_success) {
|
|
return srs_error_wrap(err, "range");
|
|
}
|
|
} else {
|
|
// unknown header name, parse util EOF.
|
|
std::string value;
|
|
if ((err = recv_token_util_eof(value)) != srs_success) {
|
|
return srs_error_wrap(err, "state");
|
|
}
|
|
srs_trace("rtsp: ignore header %s=%s", token.c_str(), value.c_str());
|
|
}
|
|
}
|
|
|
|
// for setup, parse the stream id from uri.
|
|
if (req->is_setup()) {
|
|
size_t pos = string::npos;
|
|
std::string stream_id = srs_path_basename(req->uri);
|
|
if ((pos = stream_id.find("=")) != string::npos) {
|
|
stream_id = stream_id.substr(pos + 1);
|
|
}
|
|
req->stream_id = ::atoi(stream_id.c_str());
|
|
srs_info("rtsp: setup stream id=%d", req->stream_id);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtspStack::recv_token_normal(std::string& token)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRtspTokenState state;
|
|
|
|
if ((err = recv_token(token, state)) != srs_success) {
|
|
if (srs_error_code(err) == ERROR_RTSP_REQUEST_HEADER_EOF) {
|
|
return srs_error_wrap(err, "EOF");
|
|
}
|
|
return srs_error_wrap(err, "recv token");
|
|
}
|
|
|
|
if (state != SrsRtspTokenStateNormal) {
|
|
return srs_error_new(ERROR_RTSP_TOKEN_NOT_NORMAL, "invalid state=%d", state);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtspStack::recv_token_eof(std::string& token)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRtspTokenState state;
|
|
|
|
if ((err = recv_token(token, state)) != srs_success) {
|
|
if (srs_error_code(err) == ERROR_RTSP_REQUEST_HEADER_EOF) {
|
|
return srs_error_wrap(err, "EOF");
|
|
}
|
|
return srs_error_wrap(err, "recv token");
|
|
}
|
|
|
|
if (state != SrsRtspTokenStateEOF) {
|
|
return srs_error_new(ERROR_RTSP_TOKEN_NOT_NORMAL, "invalid state=%d", state);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtspStack::recv_token_util_eof(std::string& token, int* pconsumed)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
SrsRtspTokenState state;
|
|
|
|
// use 0x00 as ignore the normal token flag.
|
|
if ((err = recv_token(token, state, 0x00, pconsumed)) != srs_success) {
|
|
if (srs_error_code(err) == ERROR_RTSP_REQUEST_HEADER_EOF) {
|
|
return srs_error_wrap(err, "EOF");
|
|
}
|
|
return srs_error_wrap(err, "recv token");
|
|
}
|
|
|
|
if (state != SrsRtspTokenStateEOF) {
|
|
return srs_error_new(ERROR_RTSP_TOKEN_NOT_NORMAL, "invalid state=%d", state);
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtspStack::recv_token(std::string& token, SrsRtspTokenState& state, char normal_ch, int* pconsumed)
|
|
{
|
|
srs_error_t err = srs_success;
|
|
|
|
// whatever, default to error state.
|
|
state = SrsRtspTokenStateError;
|
|
|
|
// when buffer is empty, append bytes first.
|
|
bool append_bytes = buf->length() == 0;
|
|
|
|
// parse util token.
|
|
for (;;) {
|
|
// append bytes if required.
|
|
if (append_bytes) {
|
|
append_bytes = false;
|
|
|
|
char buffer[SRS_RTSP_BUFFER];
|
|
ssize_t nb_read = 0;
|
|
if ((err = skt->read(buffer, SRS_RTSP_BUFFER, &nb_read)) != srs_success) {
|
|
return srs_error_wrap(err, "recv data");
|
|
}
|
|
|
|
buf->append(buffer, (int)nb_read);
|
|
}
|
|
|
|
// Try to detect and consume any RTCP frames from the buffer
|
|
while (buf->length() > 0) {
|
|
srs_error_t rtcp_err = try_consume_rtcp_frame();
|
|
|
|
if (rtcp_err == srs_success) {
|
|
// Successfully consumed an RTCP frame, continue to check for more
|
|
continue;
|
|
} else if (srs_error_code(rtcp_err) == ERROR_RTSP_NEED_MORE_DATA) {
|
|
// Need more data to complete RTCP frame, let the outer loop read more
|
|
srs_freep(rtcp_err);
|
|
append_bytes = true;
|
|
break;
|
|
} else {
|
|
// Not an RTCP frame or other error, break and try RTSP parsing
|
|
srs_freep(rtcp_err);
|
|
break;
|
|
}
|
|
}
|
|
|
|
// parse one by one.
|
|
char* start = buf->bytes();
|
|
char* end = start + buf->length();
|
|
char* p = start;
|
|
|
|
// find util SP/CR/LF, max 2 EOF, to finger out the EOF of message.
|
|
for (; p < end && p[0] != normal_ch && p[0] != SRS_RTSP_CR && p[0] != SRS_RTSP_LF; p++) {
|
|
}
|
|
|
|
// matched.
|
|
if (p < end) {
|
|
// finger out the state.
|
|
if (p[0] == normal_ch) {
|
|
state = SrsRtspTokenStateNormal;
|
|
} else {
|
|
state = SrsRtspTokenStateEOF;
|
|
}
|
|
|
|
// got the token.
|
|
int nb_token = (int)(p - start);
|
|
// trim last ':' character.
|
|
if (nb_token && p[-1] == ':') {
|
|
nb_token--;
|
|
}
|
|
if (nb_token) {
|
|
token.append(start, nb_token);
|
|
} else {
|
|
err = srs_error_new(ERROR_RTSP_REQUEST_HEADER_EOF, "EOF");
|
|
}
|
|
|
|
// ignore SP/CR/LF
|
|
for (int i = 0; i < 2 && p < end && (p[0] == normal_ch || p[0] == SRS_RTSP_CR || p[0] == SRS_RTSP_LF); p++, i++) {
|
|
}
|
|
|
|
// consume the token bytes.
|
|
srs_assert(p - start);
|
|
buf->erase((int)(p - start));
|
|
if (pconsumed) {
|
|
*pconsumed = (int)(p - start);
|
|
}
|
|
break;
|
|
}
|
|
|
|
// append more and parse again.
|
|
append_bytes = true;
|
|
}
|
|
|
|
return err;
|
|
}
|
|
|
|
srs_error_t SrsRtspStack::try_consume_rtcp_frame()
|
|
{
|
|
// Need at least 4 bytes for RTCP over TCP header: $ + channel + length
|
|
if (buf->length() < 4) {
|
|
// Not enough data, let caller read more
|
|
return srs_error_new(ERROR_RTSP_NEED_MORE_DATA, "need more data for rtcp header");
|
|
}
|
|
|
|
char* data = buf->bytes();
|
|
|
|
// Check for RTCP over TCP format: $ + channel + length(2 bytes)
|
|
if (data[0] == '$') {
|
|
uint8_t channel = (uint8_t)data[1];
|
|
uint16_t payload_length = (uint16_t(data[2]) << 8) | uint16_t(data[3]);
|
|
int total_frame_size = 4 + payload_length; // 4-byte header + payload
|
|
|
|
// Check if we have the complete frame
|
|
if (buf->length() < total_frame_size) {
|
|
// Not enough data for complete frame, let caller read more
|
|
return srs_error_new(ERROR_RTSP_NEED_MORE_DATA, "need more data for complete rtcp frame");
|
|
}
|
|
|
|
// Check if the payload is RTCP (starts at offset 4)
|
|
if (payload_length >= 8 && srs_is_rtcp((const uint8_t*)(data + 4), payload_length)) {
|
|
// This is an RTCP packet in RTSP over TCP format
|
|
srs_trace("RTSP: Consuming RTCP packet(%d), channel=%d, size=%d bytes",
|
|
(uint8_t)data[5], channel, payload_length);
|
|
buf->erase(total_frame_size);
|
|
return srs_success;
|
|
} else {
|
|
// Unknown interleaved frame, consume it anyway to avoid blocking RTSP parsing
|
|
srs_trace("RTSP: Consuming unknown interleaved frame, channel=%d, size=%d bytes",
|
|
channel, payload_length);
|
|
buf->erase(total_frame_size);
|
|
return srs_success;
|
|
}
|
|
}
|
|
|
|
// Not an interleaved frame (RTP/RTCP over TCP)
|
|
return srs_error_new(ERROR_RTSP_TOKEN_NOT_NORMAL, "not interleaved frame");
|
|
}
|