VOIP RTP RTSP 實現 Baresip 源碼分析

RTP 使用 udp 進行數據傳輸，udp 是不能保證，數據包必定能夠到達的，也不提供時序。同時還有 MTU 限制。

RTCP 用來配合 RTP 提供，傳輸報告，會話創建和退出。

一大批參考規範

 1 * [RFC 1321](https://tools.ietf.org/html/rfc1321) - The MD5 Message-Digest Algorithm
 2 * [RFC 1886](https://tools.ietf.org/html/rfc1886) - DNS Extensions to support IP version 6
 3 * [RFC 2032](https://tools.ietf.org/html/rfc2032) - RTP Payload Format for H.261 Video Streams
 4 * [RFC 2616](https://tools.ietf.org/html/rfc2616) - Hypertext Transfer Protocol -- HTTP/1.1
 5 * [RFC 2617](https://tools.ietf.org/html/rfc2617) - HTTP Authentication: Basic and Digest Access Authentication
 6 * [RFC 2782](https://tools.ietf.org/html/rfc2782) - A DNS RR for Specifying the Location of Services (DNS SRV)
 7 * [RFC 2915](https://tools.ietf.org/html/rfc2915) - The Naming Authority Pointer (NAPTR) DNS Resource Record
 8 * [RFC 3261](https://tools.ietf.org/html/rfc3261) - SIP: Session Initiation Protocol
 9 * [RFC 3263](https://tools.ietf.org/html/rfc3263) - Locating SIP Servers
10 * [RFC 3264](https://tools.ietf.org/html/rfc3264) - An Offer/Answer Model with SDP
11 * [RFC 3265](https://tools.ietf.org/html/rfc3265) - SIP-Specific Event Notification
12 * [RFC 3327](https://tools.ietf.org/html/rfc3327) - SIP Extension Header Field for Registering Non-Adjacent Contacts
13 * [RFC 3428](https://tools.ietf.org/html/rfc3428) - SIP Extension for Instant Messaging
14 * [RFC 3489](https://tools.ietf.org/html/rfc3489) - STUN - Simple Traversal of UDP Through NATs
15 * [RFC 3515](https://tools.ietf.org/html/rfc3515) - The SIP Refer Method
16 * [RFC 3550](https://tools.ietf.org/html/rfc3550) - RTP: A Transport Protocol for Real-Time Applications
17 * [RFC 3551](https://tools.ietf.org/html/rfc3551) - RTP Profile for Audio and Video Conferences with Minimal Control
18 * [RFC 3555](https://tools.ietf.org/html/rfc3555) - MIME Type Registration of RTP Payload Formats
19 * [RFC 3556](https://tools.ietf.org/html/rfc3556) - SDP Bandwidth Modifiers for RTCP Bandwidth
20 * [RFC 3581](https://tools.ietf.org/html/rfc3581) - An Extension to SIP for Symmetric Response Routing
21 * [RFC 3605](https://tools.ietf.org/html/rfc3605) - RTCP attribute in SDP
22 * [RFC 3711](https://tools.ietf.org/html/rfc3711) - The Secure Real-time Transport Protocol (SRTP)
23 * [RFC 3969](https://tools.ietf.org/html/rfc3969) - The IANA URI Parameter Registry for SIP
24 * [RFC 3994](https://tools.ietf.org/html/rfc3994) - Indication of Message Composition for Instant Messaging
25 * [RFC 4346](https://tools.ietf.org/html/rfc4346) - The TLS Protocol Version 1.1
26 * [RFC 4566](https://tools.ietf.org/html/rfc4566) - SDP: Session Description Protocol
27 * [RFC 4582](https://tools.ietf.org/html/rfc4582) - The Binary Floor Control Protocol (BFCP)
28 * [RFC 4582bis](https://tools.ietf.org/html/draft-ietf-bfcpbis-rfc4582bis-08) - The Binary Floor Control Protocol (BFCP)
29 * [RFC 4585](https://tools.ietf.org/html/rfc4585) - Extended RTP Profile for RTCP-Based Feedback
30 * [RFC 4733](https://tools.ietf.org/html/rfc4733) - RTP Payload for DTMF Digits, Telephony Tones, and Teleph. Signals
31 * [RFC 4961](https://tools.ietf.org/html/rfc4961) - Symmetric RTP / RTP Control Protocol (RTCP)
32 * [RFC 5118](https://tools.ietf.org/html/rfc5118) - SIP Torture Test Messages for IPv6
33 * [RFC 5245](https://tools.ietf.org/html/rfc5245) - Interactive Connectivity Establishment (ICE)
34 * [RFC 5389](https://tools.ietf.org/html/rfc5389) - Session Traversal Utilities for NAT (STUN)
35 * [RFC 5626](https://tools.ietf.org/html/rfc5626) - Managing Client-Initiated Connections in SIP
36 * [RFC 5761](https://tools.ietf.org/html/rfc5761) - Multiplexing RTP Data and Control Packets on a Single Port
37 * [RFC 5766](https://tools.ietf.org/html/rfc5766) - Traversal Using Relays around NAT (TURN)
38 * [RFC 5768](https://tools.ietf.org/html/rfc5768) - Indicating Support for ICE in SIP
39 * [RFC 5769](https://tools.ietf.org/html/rfc5769) - Test vectors for STUN
40 * [RFC 5780](https://tools.ietf.org/html/rfc5780) - NAT Behaviour Discovery Using STUN
41 * [RFC 6026](https://tools.ietf.org/html/rfc6026) - Correct Transaction Handling for 2xx Resp. to SIP INVITE Requests
42 * [RFC 6156](https://tools.ietf.org/html/rfc6156) - TURN Extension for IPv6
43 * [RFC 6188](https://tools.ietf.org/html/rfc6188) - The Use of AES-192 and AES-256 in Secure RTP
44 * [RFC 6455](https://tools.ietf.org/html/rfc6455) - The WebSocket Protocol
45 * [RFC 7159](https://tools.ietf.org/html/rfc7159) - JavaScript Object Notation (JSON)
46 * [RFC 7350](https://tools.ietf.org/html/rfc7350) - DTLS as Transport for STUN
47 * [RFC 7714](https://tools.ietf.org/html/rfc7714) - AES-GCM Authenticated Encryption in SRTP

 

有幾關鍵問題：

1，udp 怎麼創建

2,  聲音定時器多少時間傳輸入一次

3,  丟包問題 jitter buffer 

下面使用 Wireshark 對比 Baresip 源碼來進行分析。

/** * \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 的端口號是 RTP 的端口號加1 。

libre 源碼分析

1,  創建udp socket 非阻塞方式 

2, 初始化 epoll 註冊回調函數 udp_read()

3, 接收到數據 內核通知 epoll 調用 udp_read() 初始化 mbuf

4, call helpers 回調每個註冊的 helper

baresip 聲音相關結構體：

1 /** Audio Source parameters */
2 struct ausrc_prm { 3     uint32_t   srate;       /**< Sampling rate in [Hz] */
4     uint8_t    ch;          /**< Number of channels */
5     uint32_t   ptime;       /**< Wanted packet-time in [ms] */
6     int        fmt;         /**< Sample format (enum aufmt) */
7 };

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

默認 8000hz 1ch 10ms 16bit

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

RTP 頭信息

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

payload type 在這裏看 https://tools.ietf.org/html/rfc3551#page-32

我把經常使用的給標了紅色 PCMU PCMA G722 G729

PT encoding media type clock rate channels name (Hz) ___________________________________________________ 0 PCMU A 8,000 1
1 reserved A 2 reserved A 3    GSM         A            8,000       1
4    G723        A            8,000       1
5    DVI4        A            8,000       1
6    DVI4        A           16,000       1
7    LPC         A            8,000       1
8 PCMA A 8,000 1 9 G722 A 8,000 1
10   L16         A           44,100       2
11   L16         A           44,100       1
12   QCELP       A            8,000       1
13   CN          A            8,000       1
14   MPA         A           90,000 (see text) 15   G728        A            8,000       1
16   DVI4        A           11,025       1
17   DVI4        A           22,050       1
18 G729 A 8,000 1

 

RTP 12個字節的頭信息：

SSRC 之後就是 payload 。

開始的包：

 

剛一開始，Marker 是1 。

聲音參數配置

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

測試了基本可用，但回聲，和同步，丟包還未實現。