GStreamer RTSP Server RTP透传实现遇预滚失败问题求助

需求 #

实现RTSP Server的透传功能：一个端点接收推送的RTP数据包，另一个端点允许客户端拉流观看这些数据包。

当前实现 #

推送命令 #

gst-launch-1.0 videotestsrc ! video/x-raw,width=640,height=480,format=I420,framerate=30/1,clock-rate=90000 ! x264enc ! h264parse ! rtspclientsink location=rtsp://localhost:8554/push protocols=tcp

Record模式Media Factory管道（接收推送） #

self.set_launch("""rtph264depay name=depay0 ! queue ! intersink producer-name="rtp-source"""")

Pull模式Media Factory管道（提供拉流） #

self.set_launch("""intersrc producer-name="rtp-source" ! h264parse ! rtph264pay pt=96 config-interval=5 name=pay0""")

工厂挂载逻辑（客户端连接回调） #

def on_options_request(self, client, context):
    global routes
    print(client)
    print(context)
    if 'test' not in routes:
        print("test not in routes")
        push_factory = RecordMediaFactory()
        pull_factory = PlayMediaFactory()

        routes = {'test': {'push': None, 'pull': None}}
        routes['test']['push'] = push_factory
        routes['test']['pull'] = pull_factory

        self.get_mount_points().add_factory("/push", push_factory)
        self.get_mount_points().add_factory("/pull", pull_factory)

遇到的问题 #

拉流时preroll失败，服务器无法响应客户端的DESCRIBE请求，客户端提示“no media”和“can't get sdp”。服务器超时堆栈跟踪如下：

0:00:20.363756958 76394 0x600003645e00 DEBUG             rtspstream rtsp-stream.c:4840:on_message_sent:<GstRTSPStream@0x13d70aad0> message send complete
0:00:20.363774000 76394 0x600003645e00 INFO              rtspclient rtsp-client.c:2334:do_keepalive: keep session 0x600002044710 alive
0:00:23.701245291 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:3133:gst_rtsp_media_get_status: timeout, assuming error status
0:00:23.701312083 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:3139:gst_rtsp_media_get_status: got status 5
0:00:23.701325000 76394 0x60000367c000 WARN               rtspmedia rtsp-media.c:3933:wait_preroll: failed to preroll pipeline
0:00:23.701330375 76394 0x60000367c000 WARN               rtspmedia rtsp-media.c:4303:gst_rtsp_media_prepare: failed to preroll pipeline
0:00:23.701391208 76394 0x60000367c000 INFO               rtspmedia rtsp-media.c:4446:gst_rtsp_media_unprepare: unprepare media 0x13d606460
0:00:23.701406041 76394 0x60000367c000 INFO               rtspmedia rtsp-media.c:3469:set_target_state: set target state to NULL for media 0x13d606460
0:00:23.701416916 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:3106:gst_rtsp_media_set_status: setting new status to 1
0:00:23.701509291 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:4321:finish_unprepare: shutting down
0:00:23.701524208 76394 0x60000367c000 INFO               rtspmedia rtsp-media.c:3456:set_state: set state to NULL for media 0x13d606460
0:00:23.701991291 76394 0x6000036401e0 DEBUG              rtspmedia rtsp-media.c:3544:default_handle_message: 0x13d606460: went from PLAYING to PAUSED (pending NULL)
0:00:23.702419208 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:3074:media_streams_set_blocked: media 0x13d606460 set blocked 0
0:00:23.702437000 76394 0x60000367c000 DEBUG             rtspstream rtsp-stream.c:5523:set_blocked:<GstRTSPStream@0x13d607010> blocked: 0
0:00:23.702445333 76394 0x60000367c000 INFO               rtspmedia rtsp-media.c:4334:finish_unprepare: Removing elements of stream 0 from pipeline
0:00:23.702451458 76394 0x60000367c000 INFO              rtspstream rtsp-stream.c:4163:gst_rtsp_stream_leave_bin: stream 0x13d607010 leaving bin
0:00:23.702629125 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:3106:gst_rtsp_media_set_status: setting new status to 0
0:00:23.703029083 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:4377:finish_unprepare: removing bus watch
0:00:23.703042750 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:3720:watch_destroyed:<GstRTSPMedia@0x13d606460> source destroyed
0:00:23.703051375 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:4382:finish_unprepare: destroy source
0:00:23.703057541 76394 0x60000367c000 DEBUG              rtspmedia rtsp-media.c:4388:finish_unprepare: stop thread
0:00:23.703070541 76394 0x60000367c000 ERROR             rtspclient rtsp-client.c:1117:find_media: client 0x11d64a8f0: can't prepare media

其他尝试情况 #

• 用videotestsrc替代intersink/intersrc方案时，推流和拉流功能正常。
• 尝试过shmsink/src、手动设置caps的appsrc/sink等跨管道结构，拉流时两端都会冻结，最终超时返回503。其中appsrc方案中，由于客户端连接时才创建管道实例，无法在prepare前设置caps并推送缓冲区，因此改用intersink/src。

请教问题 #

1. 如何以最小开销在管道间传递数据？
2. 如何正确配置拉流端以转发传入的RTP数据包？（rtppassthroughpay看起来可行，但仍需解决跨Media Factory的管道通信问题）

内容的提问来源于stack exchange，提问作者Steve