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GB28181: Fix parse rtp tcp failed (#2382)
* fix parse rtp-tcp failed * fix parse rtp-tcp failed * fix gb28181 support tcp stack is setup:passive * Update push.gb28181.tcp.conf Co-authored-by: cfw <fangwei.cheng@transwarp.io> Co-authored-by: Winlin <winlin@vip.126.com>
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143
trunk/conf/push.gb28181.tcp.conf
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143
trunk/conf/push.gb28181.tcp.conf
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# push gb28181 stream to SRS.
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listen 1935;
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max_connections 1000;
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daemon off;
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srs_log_tank console;
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http_api {
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enabled on;
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listen 1985;
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}
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http_server {
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enabled on;
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listen 8080;
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}
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stats {
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network 0;
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}
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stream_caster {
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enabled on;
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caster gb28181;
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# 转发流到rtmp服务器地址与端口
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# TODO: https://github.com/ossrs/srs/pull/1679/files#r400875104
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# [stream] is VideoChannelCodecID(视频通道编码ID) for sip
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# 自动创建的道通[stream] 是‘chid[ssrc]’ [ssrc]是rtp的ssrc
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# [ssrc] rtp中的ssrc
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output rtmp://127.0.0.1:1935/live/[stream];
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# 接收设备端rtp流的多路复用端口
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listen 9000;
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# 多路复用端口类型，on为tcp，off为udp
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# 默认：off
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tcp_enable on;
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# rtp接收监听端口范围，最小值
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rtp_port_min 58200;
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# rtp接收监听端口范围，最大值
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rtp_port_max 58300;
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# 是否等待关键帧之后，再转发，
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# off:不需等待，直接转发
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# on:等第一个关键帧后，再转发
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wait_keyframe on;
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# rtp包空闲等待时间，如果指定时间没有收到任何包
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# rtp监听连接自动停止，发送BYE命令
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rtp_idle_timeout 30;
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# 是否转发音频流
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# 目前只支持aac格式，所以需要设备支持aac格式
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# on:转发音频
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# off:不转发音频，只有视频
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# *注意*!!!:flv 只支持11025 22050 44100 三种
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# 如果设备端没有三种中任何一个，转发时为自动选择一种格式
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# 同时也会将adts的头封装在flv aac raw数据中
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# 这样的话播放器为自动通过adts头自动选择采样频率
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# 像ffplay, vlc都可以，但是flash是没有声音，
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# 因为flash,只支持11025 22050 44100
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audio_enable off;
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# 是否开启rtp缓冲
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# 开启之后能有效解决rtp乱序等问题
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# tcp模式建议关闭
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jitterbuffer_enable off;
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# 服务器主机号，可以域名或ip地址
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# 也就是设备端将媒体发送的地址，如果是服务器是内外网
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# 需要写外网地址，
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# 调用api创建stream session时返回ip地址也是host
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# $CANDIDATE 是系统环境变量，从环境变量获取地址，如果没有配置，用*
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# *代表指定stats network 的网卡号地址，如果没有配置network，默认则是第0号网卡地址
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# TODO: https://github.com/ossrs/srs/pull/1679/files#r400917594
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host $CANDIDATE;
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#根据收到ps rtp包自带创建rtmp媒体通道，不需要api接口创建
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#rtmp地址参数[stream] 就是通道id 格式chid[ssrc]
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auto_create_channel off;
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sip {
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# 是否启用srs内部sip信令
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# 为on信令走srs, off 只转发ps流
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enabled on;
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# sip监听udp端口
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listen 5060;
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# SIP server ID(SIP服务器ID).
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# 设备端配置编号需要与该值一致，否则无法注册
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serial 34020000002000000001;
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# SIP server domain(SIP服务器域)
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realm 3402000000;
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# 服务端发送ack后，接收回应的超时时间，单位为秒
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# 如果指定时间没有回应，认为失败
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ack_timeout 30;
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# 设备心跳维持时间，如果指定时间内(秒）没有接收一个心跳
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# 认为设备离线
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keepalive_timeout 120;
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# 注册之后是否自动给设备端发送invite
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# on: 是 off 不是，需要通过api控制
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auto_play on;
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# 设备将流发送的端口，是否固定
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# on 发送流到多路复用端口 如9000
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# off 自动从rtp_mix_port - rtp_max_port 之间的值中
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# 选一个可以用的端口
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invite_port_fixed on;
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# 向设备或下级域查询设备列表的间隔，单位(秒)
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# 默认60秒
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query_catalog_interval 60;
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}
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}
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rtc_server {
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enabled on;
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# Listen at udp://8000
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listen 8000;
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#
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# The $CANDIDATE means fetch from env, if not configed, use * as default.
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#
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# The * means retrieving server IP automatically, from all network interfaces,
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# @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
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candidate $CANDIDATE;
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}
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vhost __defaultVhost__ {
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rtc {
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enabled on;
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bframe discard;
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}
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http_remux {
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enabled on;
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mount [vhost]/[app]/[stream].flv;
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}
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}
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@ -1062,6 +1062,8 @@ void SrsSipStack::req_invite(stringstream& ss, SrsSipRequest *req, string ip, in
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//<< "m=video " << port << " TCP/RTP/AVP 98" << SRS_RTSP_CRLF
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<< "a=recvonly" << SRS_RTSP_CRLF
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<< "a=rtpmap:96 PS/90000" << SRS_RTSP_CRLF
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<< "a=setup:passive" << SRS_RTSP_CRLF
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<< "a=connection:new" << SRS_RTSP_CRLF
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//TODO: current no support
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//<< "a=rtpmap:97 MPEG4/90000" << SRS_RTSP_CRLF
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//<< "a=rtpmap:98 H264/90000" << SRS_RTSP_CRLF
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