ps网站背景图片怎么做,如何增强网站的安全性,wordpress做官网,福利 源码 wordpress我们先看看官方规范针对TCP协议的视音频传输描述#xff1a;
实时视频点播、历史视频回放与下载的 TCP媒体传输应支持基于RTP封装的视音频PS流#xff0c;封装格式参照IETFRFC4571。
流媒体服务器宜同时支持作为TCP媒体流传输服务端和客户端。默认情况下#xff0c;前端设…我们先看看官方规范针对TCP协议的视音频传输描述
实时视频点播、历史视频回放与下载的 TCP媒体传输应支持基于RTP封装的视音频PS流封装格式参照IETFRFC4571。
流媒体服务器宜同时支持作为TCP媒体流传输服务端和客户端。默认情况下前端设备向流媒体服务器发送媒体流时前端设备应作为TCP媒体流传输客户端,流媒体服务器作为 TCP媒体流传输服务端;同级或跨级流媒体服务器间基于 TCP协议传输视频流时,媒体流的接收方宜作为TCP媒体流传输服务端。
媒体流的发送方和接收方可扩展SDP参数进行TCP媒体流传输服务端和客户端的协商,协商机制参考附录 F及IETFRFC4571的定义。
这里我们看个INVITE信令交互示例
INVITE sip:340200000013200000013402000000 SIP/2.0
Via: SIP/2.0/TCP 192.168.0.105:15060;rport;branchz9hG4bK630055772
From: sip:340200000020000000013402000000;tag562055772
To: sip:340200000013200000013402000000
Call-ID: 589055668
CSeq: 183 INVITE
Content-Type: APPLICATION/SDP
Contact: sip:34020000002000000001192.168.0.105:15060
Max-Forwards: 70
User-Agent: LiveGB28181
Subject: 34020000001320000001:0200000001,34020000002000000001:0
Content-Length: 222v0
o34020000001320000001 0 0 IN IP4 192.168.0.105
sPlay
cIN IP4 192.168.0.105
t0 0
mvideo 30076 RTP/AVP 96 97 98
arecvonly
artpmap:96 PS/90000
artpmap:97 MPEG4/90000
artpmap:98 H264/90000
y0200000001
判断媒体流走TCP还是UDP主要看这里
mvideo 30076 RTP/AVP 96 97 98
传输方式采用“RTP/AVP”标识传输层协议为 RTP over UDP采用“TCP/RTP/AVP”标识传输层协议为 RTP over TCP需要注意的是我们实际对接的时候部分厂商SDP非常随意有的甚至直接标记个tcp这让我们对接的时候很困惑。
技术实现 本文以大牛直播SDK的Android平台GB28181设备接入端为例启动GB28181完成注册、catalog等交互后Invite上来后设置媒体流通过TCP还是UDP发送出去
Override
public void ntsOnInvitePlay(String deviceId, SessionDescription session_des) {handler_.postDelayed(new Runnable() {Overridepublic void run() {// 先振铃响应下gb28181_agent_.respondPlayInvite(180, device_id_);MediaSessionDescription video_des null;SDPRtpMapAttribute ps_rtpmap_attr null;// 28181 视频使用PS打包VectorMediaSessionDescription video_des_list session_des_.getVideoPSDescriptions();if (video_des_list ! null !video_des_list.isEmpty()) {for(MediaSessionDescription m : video_des_list) {if (m ! null m.isValidAddressType() m.isHasAddress() ) {video_des m;ps_rtpmap_attr video_des.getPSRtpMapAttribute();break;}}}if (null video_des) {gb28181_agent_.respondPlayInvite(488, device_id_);Log.i(TAG, ntsOnInvitePlay get video description is null, response 488, device_id: device_id_);return;}if (null ps_rtpmap_attr) {gb28181_agent_.respondPlayInvite(488, device_id_);Log.i(TAG, ntsOnInvitePlay get ps rtp map attribute is null, response 488, device_id: device_id_);return;}Log.i(TAG,ntsOnInvitePlay, device_id: device_id_, is_tcp: video_des.isRTPOverTCP() rtp_port: video_des.getPort() ssrc: video_des.getSSRC() address_type: video_des.getAddressType() address: video_des.getAddress());long rtp_sender_handle libPublisher.CreateRTPSender(0);if ( rtp_sender_handle 0 ) {gb28181_agent_.respondPlayInvite(488, device_id_);Log.i(TAG, ntsOnInvitePlay CreateRTPSender failed, response 488, device_id: device_id_);return;}gb28181_rtp_payload_type_ ps_rtpmap_attr.getPayloadType();gb28181_rtp_encoding_name_ ps_rtpmap_attr.getEncodingName();libPublisher.SetRTPSenderTransportProtocol(rtp_sender_handle, video_des.isRTPOverUDP()?0:1);libPublisher.SetRTPSenderIPAddressType(rtp_sender_handle, video_des.isIPv4()?0:1);libPublisher.SetRTPSenderLocalPort(rtp_sender_handle, 0);libPublisher.SetRTPSenderSSRC(rtp_sender_handle, video_des.getSSRC());libPublisher.SetRTPSenderSocketSendBuffer(rtp_sender_handle, 2*1024*1024); // 设置到2MlibPublisher.SetRTPSenderClockRate(rtp_sender_handle, ps_rtpmap_attr.getClockRate());libPublisher.SetRTPSenderDestination(rtp_sender_handle, video_des.getAddress(), video_des.getPort());if ( libPublisher.InitRTPSender(rtp_sender_handle) ! 0 ) {gb28181_agent_.respondPlayInvite(488, device_id_);libPublisher.DestoryRTPSender(rtp_sender_handle);return;}int local_port libPublisher.GetRTPSenderLocalPort(rtp_sender_handle);if (local_port 0) {gb28181_agent_.respondPlayInvite(488, device_id_);libPublisher.DestoryRTPSender(rtp_sender_handle);return;}Log.i(TAG,get local_port: local_port);String local_ip_addr IPAddrUtils.getIpAddress(context_);MediaSessionDescription local_video_des new MediaSessionDescription(video_des.getType());local_video_des.addFormat(String.valueOf(ps_rtpmap_attr.getPayloadType()));local_video_des.addRtpMapAttribute(ps_rtpmap_attr);local_video_des.setAddressType(video_des.getAddressType());local_video_des.setAddress(local_ip_addr);local_video_des.setPort(local_port);local_video_des.setTransportProtocol(video_des.getTransportProtocol());local_video_des.setSSRC(video_des.getSSRC());if (!gb28181_agent_.respondPlayInviteOK(device_id_,local_video_des) ) {libPublisher.DestoryRTPSender(rtp_sender_handle);Log.e(TAG, ntsOnInvitePlay call respondPlayInviteOK failed.);return;}gb28181_rtp_sender_handle_ rtp_sender_handle;}private String device_id_;private SessionDescription session_des_;public Runnable set(String device_id, SessionDescription session_des) {this.device_id_ device_id;this.session_des_ session_des;return this;}}.set(deviceId, session_des),0);
}
接口设计如下
/***设置 RTP Sender传输协议** param rtp_sender_handle, CreateRTPSender返回值* param transport_protocol, 0:UDP, 1:TCP, 默认是UDP** return {0} if successful*/public native int SetRTPSenderTransportProtocol(long rtp_sender_handle, int transport_protocol);
以上是GB28181基于TCP协议的视音频媒体传输探究及实现感兴趣的开发者可以查看相关协议规范根据需求实现自己的业务逻辑即可。
