VOIP RTP RTSP 实现 Baresip 源码分析

RTP 使用 udp 进行数据传输，udp 是不能保证，数据包一定可以到达的，也不提供时序。同时还有 MTU 限制。

RTCP 用来配合 RTP 提供，传输报告，会话建立和退出。

一大批参考规范

 * [RFC ](https://tools.ietf.org/html/rfc1321) - The MD5 Message-Digest Algorithm
 * [RFC ](https://tools.ietf.org/html/rfc1886) - DNS Extensions to support IP version 6
 * [RFC ](https://tools.ietf.org/html/rfc2032) - RTP Payload Format for H.261 Video Streams
 * [RFC ](https://tools.ietf.org/html/rfc2616) - Hypertext Transfer Protocol -- HTTP/1.1
 * [RFC ](https://tools.ietf.org/html/rfc2617) - HTTP Authentication: Basic and Digest Access Authentication
 * [RFC ](https://tools.ietf.org/html/rfc2782) - A DNS RR for Specifying the Location of Services (DNS SRV)
 * [RFC ](https://tools.ietf.org/html/rfc2915) - The Naming Authority Pointer (NAPTR) DNS Resource Record
 * [RFC ](https://tools.ietf.org/html/rfc3261) - SIP: Session Initiation Protocol
 * [RFC ](https://tools.ietf.org/html/rfc3263) - Locating SIP Servers
 * [RFC ](https://tools.ietf.org/html/rfc3264) - An Offer/Answer Model with SDP
 * [RFC ](https://tools.ietf.org/html/rfc3265) - SIP-Specific Event Notification
 * [RFC ](https://tools.ietf.org/html/rfc3327) - SIP Extension Header Field for Registering Non-Adjacent Contacts
 * [RFC ](https://tools.ietf.org/html/rfc3428) - SIP Extension for Instant Messaging
 * [RFC ](https://tools.ietf.org/html/rfc3489) - STUN - Simple Traversal of UDP Through NATs
 * [RFC ](https://tools.ietf.org/html/rfc3515) - The SIP Refer Method
 * [RFC ](https://tools.ietf.org/html/rfc3550) - RTP: A Transport Protocol for Real-Time Applications
 * [RFC ](https://tools.ietf.org/html/rfc3551) - RTP Profile for Audio and Video Conferences with Minimal Control
 * [RFC ](https://tools.ietf.org/html/rfc3555) - MIME Type Registration of RTP Payload Formats
 * [RFC ](https://tools.ietf.org/html/rfc3556) - SDP Bandwidth Modifiers for RTCP Bandwidth
 * [RFC ](https://tools.ietf.org/html/rfc3581) - An Extension to SIP for Symmetric Response Routing
 * [RFC ](https://tools.ietf.org/html/rfc3605) - RTCP attribute in SDP
 * [RFC ](https://tools.ietf.org/html/rfc3711) - The Secure Real-time Transport Protocol (SRTP)
 * [RFC ](https://tools.ietf.org/html/rfc3969) - The IANA URI Parameter Registry for SIP
 * [RFC ](https://tools.ietf.org/html/rfc3994) - Indication of Message Composition for Instant Messaging
 * [RFC ](https://tools.ietf.org/html/rfc4346) - The TLS Protocol Version 1.1
 * [RFC ](https://tools.ietf.org/html/rfc4566) - SDP: Session Description Protocol
 * [RFC ](https://tools.ietf.org/html/rfc4582) - The Binary Floor Control Protocol (BFCP)
 * [RFC 4582bis](https://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4582bis-08) - The Binary Floor Control Protocol (BFCP)
 * [RFC ](https://tools.ietf.org/html/rfc4585) - Extended RTP Profile for RTCP-Based Feedback
 * [RFC ](https://tools.ietf.org/html/rfc4733) - RTP Payload for DTMF Digits, Telephony Tones, and Teleph. Signals
 * [RFC ](https://tools.ietf.org/html/rfc4961) - Symmetric RTP / RTP Control Protocol (RTCP)
 * [RFC ](https://tools.ietf.org/html/rfc5118) - SIP Torture Test Messages for IPv6
 * [RFC ](https://tools.ietf.org/html/rfc5245) - Interactive Connectivity Establishment (ICE)
 * [RFC ](https://tools.ietf.org/html/rfc5389) - Session Traversal Utilities for NAT (STUN)
 * [RFC ](https://tools.ietf.org/html/rfc5626) - Managing Client-Initiated Connections in SIP
 * [RFC ](https://tools.ietf.org/html/rfc5761) - Multiplexing RTP Data and Control Packets on a Single Port
 * [RFC ](https://tools.ietf.org/html/rfc5766) - Traversal Using Relays around NAT (TURN)
 * [RFC ](https://tools.ietf.org/html/rfc5768) - Indicating Support for ICE in SIP
 * [RFC ](https://tools.ietf.org/html/rfc5769) - Test vectors for STUN
 * [RFC ](https://tools.ietf.org/html/rfc5780) - NAT Behaviour Discovery Using STUN
 * [RFC ](https://tools.ietf.org/html/rfc6026) - Correct Transaction Handling for 2xx Resp. to SIP INVITE Requests
 * [RFC ](https://tools.ietf.org/html/rfc6156) - TURN Extension for IPv6
 * [RFC ](https://tools.ietf.org/html/rfc6188) - The Use of AES-192 and AES-256 in Secure RTP
 * [RFC ](https://tools.ietf.org/html/rfc6455) - The WebSocket Protocol
 * [RFC ](https://tools.ietf.org/html/rfc7159) - JavaScript Object Notation (JSON)
 * [RFC ](https://tools.ietf.org/html/rfc7350) - DTLS as Transport for STUN
 * [RFC ](https://tools.ietf.org/html/rfc7714) - AES-GCM Authenticated Encryption in SRTP

有几关键问题：

1，udp 怎么建立

2,  声音定时器多少时间传输入一次

3,  丢包问题 jitter buffer

下面使用 Wireshark 对比 Baresip 源码来进行分析。

/**
 * \page GenericAudioStream Generic Audio Stream
 *
 * Implements a generic audio stream. The application can allocate multiple
 * instances of a audio stream, mapping it to a particular SDP media line.
 * The audio object has a DSP sound card sink and source, and an audio encoder
 * and decoder. A particular audio object is mapped to a generic media
 * stream object. Each audio channel has an optional audio filtering chain.
 *
 *<pre>
 *            write  read
 *              |    /|\
 *             \|/    |
 * .------.   .---------.    .-------.
 * |filter|<--|  audio  |--->|encoder|
 * '------'   |         |    |-------|
 *            | object  |--->|decoder|
 *            '---------'    '-------'
 *              |    /|\
 *              |     |
 *             \|/    |
 *         .------. .-----.
 *         |auplay| |ausrc|
 *         '------' '-----'
 *</pre>
 */
/**
 * Audio transmit/encoder
 *
 *
 \verbatim

 Processing encoder pipeline:

 .    .-------.   .-------.   .--------.   .--------.   .--------.
 |    |       |   |       |   |        |   |        |   |        |
 |O-->| ausrc |-->| aubuf |-->| resamp |-->| aufilt |-->| encode |---> RTP
 |    |       |   |       |   |        |   |        |   |        |
 '    '-------'   '-------'   '--------'   '--------'   '--------'

 \endverbatim
 *
 */

/**
 * Audio receive/decoder
 *
 \verbatim

 Processing decoder pipeline:

       .--------.   .-------.   .--------.   .--------.   .--------.
 |\    |        |   |       |   |        |   |        |   |        |
 | |<--| auplay |<--| aubuf |<--| resamp |<--| aufilt |<--| decode |<--- RTP
 |/    |        |   |       |   |        |   |        |   |        |
       '--------'   '-------'   '--------'   '--------'   '--------'

 \endverbatim
 */ 

RTCP 的端口号是 RTP 的端口号加1 。

libre 源码分析

1,  建立udp socket 非阻塞方式

2, 初始化 epoll 注册回调函数 udp_read()

3, 接收到数据 内核通知 epoll 调用 udp_read() 初始化 mbuf

4, call helpers 回调每一个注册的 helper

baresip 声音相关结构体：

 /** Audio Source parameters */
 struct ausrc_prm {
     uint32_t   srate;       /**< Sampling rate in [Hz]      */
     uint8_t    ch;          /**< Number of channels         */
     uint32_t   ptime;       /**< Wanted packet-time in [ms] */
     int        fmt;         /**< Sample format (enum aufmt) */
 };

打印的日志：alsa: reset: srate=8000, ch=1, num_frames=160, pcmfmt=S16_LE

默认 8000hz 1ch 10ms 16bit

8000*1*16/8/1000*10 = 160

RTP 头信息

https://tools.ietf.org/html/rfc3550#section-5.1

payload type 在这里看 https://tools.ietf.org/html/rfc3551#page-32

我把常用的给标了红色 PCMU PCMA G722 G729

PT   encoding    media type  clock rate   channels
     name                    (Hz)
___________________________________________________
0    PCMU        A            8,000       1
    reserved    A
    reserved    A
    GSM         A            ,
    G723        A            ,
    DVI4        A            ,
    DVI4        A           ,
    LPC         A            ,
8    PCMA        A            8,000       1
9    G722        A            8,000       1
   L16         A           ,
   L16         A           ,
   QCELP       A            ,
   CN          A            ,
   MPA         A           ,       (see text)
   G728        A            ,
   DVI4        A           ,
   DVI4        A           ,
18   G729        A            8,000       1

RTP 12个字节的头信息：

SSRC 以后就是 payload 。

开始的包：

刚一开始，Marker 是1 。

声音参数配置

8bit sample rate 8000 2 channel ，间隔 20ms 。

测试了基本可用，但回声，和同步，丢包还未实现。