[issues] webrtc 接入SRS丢包率不正确问题

目录
[issues] webrtc 接入SRS丢包率不正确问题
原因和解决方法
srs增加rtx
SDP协商
构建RTX包

[issues] webrtc 接入SRS丢包率不正确问题

---

原因和解决方法

直接原因: SRS暂不支持RTX通道发送nack重传包，重传包以media形式发送，sdk没有区分开来，nack重传包未计入丢包统计，得到的丢包率远低于实际丢包率

解决方法:

1.StreamStatisticianImpl::UpdateOutOfOrder->IsRetransmitOfOldPacket 优化计算重传包函数，可以根据重传包达到时间与RTT比较，将时差长的包计入丢包，粗略计算丢包率。

2.srs 增加rtx 通道发送nack，需要修改SDP协商部分，和rtx编码封包nack重传包.

srs增加rtx

这里简单过来拉流端增加方法

RTX在webrtc SDK里面视频是默认打开RTX的[M88]，SRS没有支持，需要先在SRS SDP协商里面解析和分配rtx相关字段.

SDP协商

SrsRtcConnection::negotiate_play_capability 里面分配 rtx_ssrc_, rtx_pt, rtx_apt。其他还有 FID ssrc_groups

    // TODO: FIXME: set audio_payload rtcp_fbs_,

    // according by whether downlink is support transport algorithms.

    // TODO: FIXME: if we support downlink RTX, MUST assign rtx_ssrc_, rtx_pt, rtx_apt

    // not support rtx

    vector<SrsMediaPayloadType> rtx_pts = remote_media_desc.find_media_with_encoding_name("rtx");

    if (true) {

        //srs_freep(track->rtx_);

        //track->rtx_ssrc_ = 0;

        track->rtx_ssrc_ = SrsRtcSSRCGenerator::instance()->generate_ssrc();

        for (size_t i = 0; i < rtx_pts.size(); i++) {

            SrsMediaPayloadType rtx_pt = rtx_pts.at(i);

            uint8_t pt = ::atol(rtx_pt.format_specific_param_.substr(4, 3).c_str());

            if (track->media_->pt_ == pt) {

                if (!track->rtx_) {

                    track->rtx_ = new SrsCodecPayload();

                }

                track->rtx_->pt_of_publisher_ = track->rtx_->pt_;

                track->rtx_->pt_ = rtx_pt.payload_type_;

                track->rtx_->sample_ = rtx_pt.clock_rate_;

                ((SrsRtxPayloadDes*)(track->rtx_))->apt_ = pt;//::atol(rtx_pt.format_specific_param_.c_str());

                break;

            }

        }

    }

sdp 协商调试后，sdk 收到的sdp rtx部分大致长这个样子：

a=rtpmap:101 rtx/90000

a=fmtp:101 apt=100

a=ssrc-group:FID 5333335 5333336

a=ssrc:5333335 cname:15817311631_8010

a=ssrc:5333335 msid:stream_id video_label

a=ssrc:5333335 mslabel:stream_id

a=ssrc:5333335 label:video_label

a=ssrc:5333336 cname:15817311631_8010

a=ssrc:5333336 msid:stream_id video_label

a=ssrc:5333336 mslabel:stream_id

a=ssrc:5333336 label:video_label

a=candidate:0 1 udp 2130706431 192.168.6.54 8000 typ host generation 0

构建RTX包

rtx构建很简单，新定义个 SrsRtpRtxPayload, rtp头重写下type 序列号等， payload copy一下 头两个字节写media 的序列号， 对照着SDK里面加，打印下hex调试下即可。

// RTX Payload.

class SrsRtpRtxPayload : public ISrsRtpPayloader

{

public:

    char* payload;

    int size;

    uint16_t sequence_number;

public:

    SrsRtpRtxPayload();

    virtual ~SrsRtpRtxPayload();

// interface ISrsRtpPayloader

public:

    virtual uint64_t nb_bytes();

    virtual srs_error_t encode(SrsBuffer* buf);

    virtual srs_error_t decode(SrsBuffer* buf);

    virtual ISrsRtpPayloader* copy();

};

SrsRtpPacket* SrsRtcSendTrack::build_rtx_packet(

    SrsRtpPacket* packet) {

    SrsRtpPacket* rtx_packet;

    SrsRtcTrackDescription* track_desc = get_rtc_track_desc();

    SrsRtxPayloadDes* rtx_desc = (SrsRtxPayloadDes*)(track_desc->rtx_);

    uint8_t pt = packet->header.get_payload_type();

    uint8_t ppt = rtx_desc->pt_;

    uint8_t apt = rtx_desc->apt_;

    rtx_packet = packet->copy_with_no_payload();

    rtx_packet->header = packet->header;

    rtx_packet->header.set_payload_type(ppt);

    static uint16_t sequence_number_rtx_ = 1234; //only test

    rtx_packet->header.set_sequence(sequence_number_rtx_++);

    rtx_packet->header.set_ssrc(track_desc->rtx_ssrc_);

    char buf[kRtpPacketSize];

    SrsBuffer* stream = new SrsBuffer(buf, sizeof(buf));

    SrsAutoFree(SrsBuffer, stream);

    packet->payload()->encode(stream);

    SrsRtpRtxPayload* rtx_payload = new SrsRtpRtxPayload();

    rtx_payload->decode(stream);

    rtx_payload->sequence_number = packet->header.get_sequence();

    rtx_packet->set_payload(rtx_payload, SrsRtspPacketPayloadTypeRTX);

    rtx_packet->retransmission_ = true;

    return rtx_packet;

}

[issues] webrtc 接入SRS丢包率不正确问题的相关教程结束。