RTCWeb及WebRTC研究報(bào)告.doc

RTCWeb WebRTC 研究報(bào)告 IETF RTCWeb 草案研究報(bào)告 第頁 目錄目錄 術(shù)語和縮略語術(shù)語和縮略語 4 前言前言 4 1RTCWEB 與與 WEBRTC 綜述綜述 4 1 1RTCWEB與 WEBRTC 簡介 4 1 2RTCWEB基本結(jié)構(gòu) 4 1 2 1瀏覽器模型 4 1 2 2RTCWeb 梯形模型 4 1 3WEBRTC 基本結(jié)構(gòu) 4 2RTCWEB 與與 WEBRTC 規(guī)范規(guī)范 4 2 1RTCWEB 規(guī)范 4 2 1 1數(shù)據(jù)傳輸 4 2 1 2數(shù)據(jù)成幀 4 2 1 3數(shù)據(jù)格式 4 2 1 4連接管理 4 2 1 5展現(xiàn)和控制 4 2 1 6本地系統(tǒng)支持 4 2 2WEBRTC 規(guī)范 4 2 3關(guān)于 RTP 4 2 3 1網(wǎng)絡(luò)拓?fù)浣Y(jié)構(gòu) 4 3 1 1RTP 會話的信號要求 4 3 1 2WebRTC 使用的 RTP 核心協(xié)議 4 3 1 3RTP 的優(yōu)化 4 3 1 4RTP 的擴(kuò)展 4 4各總司對各總司對 RTCWEB 及及 WEBRTC 規(guī)范的提案規(guī)范的提案 4 4 1ALVESTRAND提案 4 4 2CBRAN提案 4 4 3IBC 提案 4 4 4JENNINGS提案 4 4 5JESUP提案 4 4 6JOHNSTON提案 4 4 7KAPLAN提案 4 4 8LENNOX提案 4 4 9NANDAKUMAR提案 4 4 10PARTHA提案 4 4 11ROSENBERG提案 4 4 12JOHOSTON提案 4 4 13WENGER提案 4 4 14針對同一問題不同公司提案的比較 4 4 15公司提案小結(jié) 4 5RTCWEB 與與 WEBRTC 規(guī)范的異同點(diǎn)分析規(guī)范的異同點(diǎn)分析 4 5 1相同點(diǎn) 4 IETF RTCWeb 草案研究報(bào)告 第頁 5 2不同點(diǎn) 4 6總結(jié)總結(jié) 4 主要參考文獻(xiàn)主要參考文獻(xiàn) 4 IETF RTCWeb 草案研究報(bào)告 第頁 術(shù)語和縮略語術(shù)語和縮略語 縮寫縮寫全稱解釋參考 RTCWebReal Time Communication in WEB browsers 基于網(wǎng)頁瀏覽器的實(shí)時通信 http tools i etf org wg rtc web APIApplication Programming Interface 應(yīng)用程序編程接口 a specification of a set of calls and events usually tied to a programming language or an abstract formal specification such as WebIDL with its defined semantics draft ietf rtcweb overview 02 ICEInternet Communications Engine 一種面向?qū)ο蟮闹虚g件平臺 為構(gòu)建 面向?qū)ο蟮目蛻?服務(wù)器應(yīng)用提供了 工具 API 和庫支持 Ice 應(yīng)用適合在 異種環(huán)境中使用 客戶和服務(wù)器可以 用不同的編程語言編寫 可以運(yùn)行在 不同的操作系統(tǒng)和機(jī)器架構(gòu)上 并且 可以使用多種網(wǎng)絡(luò)技術(shù)進(jìn)行通信 無 論部署環(huán)境如何 這些應(yīng)用的源碼都 是可移植的 http baike b 64590 htm 3 the vision is that the browser will have the functions that are needed for a Web application working in conjunction with its backend servers to implement these functions This means that two vital interfaces need specification 1 The protocols that browsers talk to each other without any intervening servers 2 and the APIs that are offered for a Javascript application to take advantage of the browser s functionality IETF RTCWeb 草案研究報(bào)告 第頁 1 2 2RTCWeb 梯形模型梯形模型 1 2 2 1RTCWeb 梯形模型圖梯形模型圖 圖圖 12 Browser RTC Trapezoid 1 2 2 2模型說明模型說明 On this drawing the critical part to note is that the media path low path goes directly between the browsers so it has to be Conformant 符合 to the specifications of the RTCWEB protocol suite the signalling path high path goes via servers that can modify translate or massage the signals as needed 這個模型中的各個部分分別需要一些功能組 在第二部分中我們分析在 RTCWeb 中所 需要的基本功能和相關(guān)的技術(shù) IETF RTCWeb 草案研究報(bào)告 第頁 1 3 WebRTC 基本結(jié)構(gòu)基本結(jié)構(gòu) 圖圖 13 WebRTC 體系結(jié)構(gòu)圖體系結(jié)構(gòu)圖 1 視頻 WebRTC 的視頻部分 包含采集 編解碼 I420 VP8 加密 媒體文件 圖像處理 顯示 網(wǎng)絡(luò)傳輸與流控 RTP RTCP 等功能 2 音頻 WebRTC 的音頻部分 包含設(shè)備 編解碼 iLIBC iSAC G722 PCM16 RED AVT NetEQ 加密 聲音文件 聲音處理 聲音 輸出 音量控制 音視頻同步 網(wǎng)絡(luò)傳輸與流控 RTP RTCP 等功能 IETF RTCWeb 草案研究報(bào)告 第頁 2 RTCWEB 與與 WEBRTC 規(guī)范規(guī)范 2 1 RTCWEB 規(guī)范規(guī)范 在 IETF 提出的 RTCWEB 規(guī)范中 對應(yīng)模型的不同層次提出了一些基本的功能需求 下面我們就來看看這些功能組 2 1 1 數(shù)據(jù)傳輸數(shù)據(jù)傳輸 建立安全的鏈接 擁塞控制 帶寬分配等 Data transport TCP UDP and the means to securely set up connections between entities as well as the functions for deciding when to send data Congestion management bandwidth estimation and so on Data transport refers to the sending and receiving of data over the network interfaces the choice of network layer addresses at each end of the communication and the interaction with any intermediate entities that handle the data but do not modify it such as TURN relays It includes necessary functions for congestion control When not to send data ICE is required for all media paths that use UDP in addition to the ability to pass NAT boxes ICE fulfils the need for guaranteeing that the media path is going to an UDP port that is willing to receive the data 2 1 2 數(shù)據(jù)成幀數(shù)據(jù)成幀 規(guī)定數(shù)據(jù)包的格式 Data framing RTP and other data formats that serve as containers and their functions for data confidentiality and integrity The format for media transport is RTP RFC3550 Implementation of SRTP RFC3711 is required for all implementations RTP 的使用 見文檔 2 3 部分 2 1 3 數(shù)據(jù)格式數(shù)據(jù)格式 系統(tǒng)中數(shù)據(jù)的編譯碼器說明 格式說明 功能性說明 Data formats Codec specifications format specifications and functionality specifications for the data passed between systems Audio and video codecs as well as formats for data and document sharing belong in this category In order to make use of data formats a way to describe them a session description is needed The intent of this specification is to allow each communications event to use the data formats that are best suited for that particular instance where a format is supported by both sides of the connection However a minimum standard is greatly helpful in order to ensure that communication can be achieved This document specifies a minimum baseline that will be supported by all implementations of this specification and leaves further codecs to be included at the will of the implementor IETF RTCWeb 草案研究報(bào)告 第頁 2 1 4 連接管理連接管理 會話的連接建立 分析數(shù)據(jù)格式 改變會話過程中的數(shù)據(jù)格式 Connection management Setting up connections agreeing on data formats changing data formats during the duration of a call SIP and Jingle XMPP belong in this category 連接管理部分包括多方面的規(guī)范和原則 不同公司也提出了一些相關(guān)的提案 我們將 在第三部分中討論 2 1 5 展現(xiàn)和控制展現(xiàn)和控制 包括地面控制 屏幕布局 聲控 圖像切換及其他等功能 Presentation and control What needs to happen in order to ensure that interactions behave in a non surprising manner This can include floor control screen layout voice activated image switching and other such functions where part of the system require the cooperation between parties Cisco Tandberg s TIP was one attempt at specifying this functionality The most important part of control is the user s control over the browser s interaction with input output devices and communications channels It is important that the user have some way of figuring out where his audio video or texting is being sent for what purported reason and what guarantees are made by the parties that form part of this control channel This is largely a local function between the browser the underlying operating system and the user interface this is being worked on as part of the W3C API effort 2 1 6 本地系統(tǒng)支持本地系統(tǒng)支持 Local system support functions These are things that need not be specified uniformly 一律 地 because each participant may choose to do these in a way of the participant s choosing without affecting the bits on the wire in a way that others have to be cognizant of Examples in this category 類別 include echo cancellation some forms of it local authentication and authorization mechanisms OS access control and the ability to do local recording of conversations 對于本地系統(tǒng) 有一些功能的質(zhì)量強(qiáng)烈地影響了用戶體驗(yàn) 但確切的算法并不需要協(xié) 調(diào) 整個系統(tǒng)的定義可能需要指定整個系統(tǒng)對于那些有用的設(shè)備需要哪些特點(diǎn) 但并不需 要要求這些設(shè)備按照一個固定的方式執(zhí)行 2 2 WEBRTC 規(guī)范規(guī)范 WEBRTC 主要是對 APIs 進(jìn)行了規(guī)范 規(guī)范中定義了音視頻的步驟 Representing a multimedia stream video audio or both from local devices video cameras microphones Web cams or from prerecorded files provided by the user Recording such streams locally Connecting to remote peers using NAT traversal technologies such as ICE STUN and TURN Sending the locally produced streams to remote peers and receiving streams from remote peers Displaying such streams both the locally produced ones and the remotely obtained ones locally using the video or audio elements Sending arbitrary data to remote peers IETF RTCWeb 草案研究報(bào)告 第頁 WEBRTC 具體對 Stream API p2p connection API 及垃圾回收進(jìn)行了規(guī)范 具體參照 http dev w3 org 2011 webrtc editor webrtc html WEBRTC 開發(fā)團(tuán)隊(duì)對 W3C 標(biāo)準(zhǔn)仍在開發(fā)中 所以規(guī)范隨時會有大范圍的改動 2 3 關(guān)于關(guān)于 RTP 由于對實(shí)現(xiàn) RTC IETF 和 W3C 是合作的 在 IETF 的規(guī)范文檔中也就 WEBRTC 項(xiàng) 目使用 RTP 協(xié)議進(jìn)行了一個建議和規(guī)范 網(wǎng)絡(luò)實(shí)時通信 WebRTC 框架 旨在通過使用音頻 視頻 合作 游戲等提供直接 的互動豐富的瀏覽器之間的通信 接下來主要介紹實(shí)時傳輸協(xié)議 RTP 是如何在 webRTC 這個框架中使用的 參考 draft ietf rtcweb rtp usage 01 2 3 1 網(wǎng)絡(luò)拓?fù)浣Y(jié)構(gòu)網(wǎng)絡(luò)拓?fù)浣Y(jié)構(gòu) 1 Point to Point A B 圖圖 3 13 1 PointPoint toto PointPoint 一個用戶對應(yīng)一個程序會很常見 2 Multi unicast A B v v C 圖圖 3 23 2 Multi unicastMulti unicast 用于小型會議 每一個人都發(fā)送個人單播的 RTP UDP 流向其他參與者 很 占帶寬 eg 如果有 5 個與會者 一個消息就要復(fù)制 4 份 這種拓?fù)浣Y(jié)構(gòu)很有可 能作為一個單獨(dú)的 RTP 會話 3RTP Mixer with Only Unicast Paths IETF RTCWeb 草案研究報(bào)告 第頁 A B Mixer C D 圖圖 3 33 3 RTPRTP MixerMixer withwith OnlyOnly UnicastUnicast PathsPaths An RTP mixer Figure 3 is a centralised point that selects or mixes content in a conference to optimise the RTP session so that each end point only needs connect to one entity the mixer The mixer also reduces the bit rate needs as the media sent from the mixer to the end point can be optimised in different ways These optimizations include methods like only choosing media from the currently most active speaker or mixing together audio so that only one audio streamis required in stead of 3 in the depicted scenario The downside of the mixer is that someone is required to provide the actual mixer 4 RTP Translator Relay with Only Unicast Paths A B Translator C D 圖圖 3 43 4 RTPRTP TranslatorTranslator Relay Relay withwith OnlyOnly UnicastUnicast PathsPaths 5 Translator towards Legacy end point A Translator B 圖圖 3 53 5 TranslatorTranslator towardstowards LegacyLegacy end pointend point 3 1 1 RTP 會話的信號要求會話的信號要求 1 Signalling for RTP sessions 2 Lack of Signalling for Payload Format Changes IETF RTCWeb 草案研究報(bào)告 第頁 3 1 2WebRTC 使用的使用的 RTP 核心協(xié)議 核心協(xié)議 1 RTP 和 RTCP 實(shí)時傳輸協(xié)議 REQUIRED 兩部分組成 數(shù)據(jù)傳輸協(xié)議和 RTP 控制協(xié)議 RTCP 2 RTP SAVPF 安全方面的協(xié)議 REQUIRED 3 1 3RTP 的優(yōu)化的優(yōu)化 1 RTP 和 RTCP 的復(fù)用 RPT 數(shù)據(jù)包和控制包在一個端口的復(fù)用是 REQUIRED 的 2 縮減 RTCP 的尺寸 REQUIRED 3 對稱的 RTP RTCP 協(xié)議 發(fā)包和接收包的 IP 地址和端口數(shù)量是相同的 REQUIRED 3 1 4RTP 的擴(kuò)展的擴(kuò)展 1 會議擴(kuò)展 OPTIONAL 2 頭擴(kuò)展 REQUIRED 3 快速同步擴(kuò)展 RECOMMENDED IETF RTCWeb 草案研究報(bào)告 第頁 4 各總司對各總司對 RTCWEB 及及 WEBRTC 規(guī)范的提案規(guī)范的提案 4 1 Alvestrand 提案提案 提案文件 draft alvestrand rtcweb congestion 01 涉及問題 RTCWeb Data transport 擁塞控制 主要內(nèi)容 兩種擁塞控制算法 接收端控制與發(fā)送端控制 內(nèi)容概要 There are two congestion control algorithms that together are seen to give reasonable performance and reasonable not perfect bandwidth sharing with other conferences and with TCP using applications that share the same links The two congestion control algorithms are sender side control algorithm and receiver side control algorithm The receive side algorithm can be further decomposed into three parts an arrival time filter an over use detector and a remote rate control An additional congestion controller resides at the sending side It bases its decisions on the round trip time packet loss and available bandwidth estimates transmitted from the receiving side Together sender side control and receiver side control implement the congestion control algorithm 4 2 Cbran 提案提案 提案文件 draft cbran rtcweb media 00 draft cbran rtcweb data 00 涉及問題 RTCWeb Data framing 實(shí)時的多媒體數(shù)據(jù)和非多媒體數(shù)據(jù)的傳輸協(xié)議和要求 主要內(nèi)容 These two documents outline that the real time media and non media data transport protocols and requirements for RTC Web client applications 提出方法 對于多媒體數(shù)據(jù)采用的是 拓展的拓展的 RTP RTCP 對于非多媒體數(shù)據(jù)采用的提議 每個應(yīng)用程序數(shù)據(jù)報(bào)發(fā)送一個字節(jié)的頭 以幫 助解復(fù)用的問題 合并后的數(shù)據(jù)報(bào)和頭通過 UDP 或 DTLS 發(fā)送給接收方 接收 器發(fā)送一個確認(rèn)它收到的每一個數(shù)據(jù)包 通過接收方向發(fā)送方發(fā)送 ack 的方法 用 TFRC SP RFC4828 計(jì)算出可使用的最大帶寬 從而限制發(fā)送方發(fā)送數(shù)據(jù)所 使用的帶寬 提案文件 draft cbran rtcweb codec 01 涉及問題 WebRTC 編譯碼器需求 主要內(nèi)容 This document outlines the codec and media processing requirements for WebRTC client application and endpoint devices 內(nèi)容概要 Audio codec requirements Video codec requirement WebRTC client requirements 提案文件 draft cbran rtcweb nat 02 IETF RTCWeb 草案研究報(bào)告 第頁 涉及問題 WebRTC NAT 主要內(nèi)容 This document outlines the network address translation NAT traversal requirements and for WebRTC client applications 內(nèi)容概要 Connection Management Requirements NAT Traversal Requirements Data Transmission Requirements IPv4 to IPv6 Transition Requirements Legacy Phone System Interoperability Requirements Connection Management Mechanism ICE as a Consent Mechanism Native ICE Support STUN Configuration 提案文件 draft cbran rtcweb negotiation 00 涉及問題 RTCWeb Connection management 協(xié)商與信令 主要內(nèi)容 This document outlines the negotiation and signaling protocols for RTC Web client application implementation 內(nèi)容概要 Negotiation Requirements The web browser MUST implement ICE such that adheres to the RTC Web NAT draft I D cbran jennings rtc web nat Signaling Protocol Requirements a small subset of SIP will be REQUIRED for all RTC WEB client application implementations In addition to the subset of SIP specification RFC3261 RTC WEB client application implementations will be REQUIRED to support DNS resolutions as specified in RFC3263 and the offer answer model with SDP as specified in RFC3264 提案文件 draft cbran rtcweb protocols 00 涉及問題 RTCWeb communication protocols 主要內(nèi)容 This document outlines that the communication protocols for realizing RTCWeb functionality within applications such as web browsers This document also proposes a set of application programming interface API requirements for controlling the protocol stack 內(nèi)容概要 這是一個綜合性的文檔 包括了之前的五個文檔的主要內(nèi)容和一些新內(nèi)容 從 Protocol Requirements API Requirements Legacy VoIP Interoperability 幾個方 面介紹了 RTCWEB 協(xié)議 4 3 IBC 提案提案 提案文件 draft ibc rtcweb sip websocket 00 涉及問題 RTCWeb WebSocket Transport for SIP 主要內(nèi)容 This document specifies a WebSocket subprotocol for a new transport in SIP Session Initiation Protocol 內(nèi)容概要 The WebSocket protocol enables two way realtime communication between clients typically web based applications and servers The main goal of this specification is to integrate the SIP protocol within web applications 使用 WebSocket 可以讓客戶端與服務(wù)器端通過 socket 端口來傳遞數(shù)據(jù) 這樣做 的好處是可以實(shí)現(xiàn)數(shù)據(jù)推送技術(shù) 服務(wù)器端不再是被動堤等待客戶端發(fā)出的 IETF RTCWeb 草案研究報(bào)告 第頁 請求 只要客戶端有一個被打開的 socket 與服務(wù)器端建立了一次連接之后 服 務(wù)器端就可以在需要的時候 主動地將數(shù)據(jù)推送到客戶端 不再需要輪詢客戶 端的請求 直到客戶端顯示關(guān)閉這個連接 4 4Jennings 提案提案 提案文件 draft jennings rtcweb signaling 01 涉及問題 RTCWeb Connection management ROAP RTCWeb Offer Answer Protocol 主要內(nèi)容 This specification defines a protocol that allows an RTCWeb browser to exchange information to control the set up of media to another browser or device 內(nèi)容概要 該提案定義了用于建立 RTCWeb 瀏覽器之間媒體協(xié)商 媒體會話建立的 ROAP 協(xié)議 該協(xié)議要求發(fā)送方和接收方先后發(fā)送 OFFER 發(fā)送方發(fā)出請求 ANSWER 接收方確認(rèn)收到 OFFER OK 發(fā)送方確認(rèn)收到 ANSWER 除此之 外還有 SHUTDOWN ERROR 等 這些信息都將通過可靠的傳輸協(xié)議被攜帶發(fā) 送 因此 ROAP 也是可靠的 在這個過程中有兩種狀態(tài) session state and request state 應(yīng)答者必須攜帶與發(fā)送者相同的狀態(tài) 提案文件 draft jennings rtcweb signaling gateway 00 涉及問題 RTCWeb Connection management Gateway between ROAP and SIP 主要內(nèi)容 This document proposes behavior of a RTCWeb signaling gateway for mapping message representations between RTCWeb Offer Answer Protocol ROAP scheme and native SIP messaging scheme 內(nèi)容概要 該提案介紹了 RTCWeb 信令網(wǎng)關(guān) 通過該網(wǎng)關(guān) RTCWeb 瀏覽器可以與使用 SIP 協(xié)議的通信終端建立會話 在會話中插入多媒體信息 結(jié)束會話 同時 提案規(guī)定了該網(wǎng)關(guān)如何處理 SIP 請求 SIP 響應(yīng)和 Web 消息 還有一些 limitations 4 5Jesup 提案提案 提案文件 draft jesup rtcweb data 01 涉及問題 RTCWeb Datagram Connection 主要內(nèi)容 This document provides Requirement and use cases for both unreliable and reliable peer to peer datagram base channel provide an overview of the pro and cons of the different proposed solutions and finally analyze in more detail the SCTP based solution 內(nèi)容概要 該文檔提出了 the requirements for P2P data connections between two browsers 之 后 列出了在可靠傳輸和不可靠傳輸兩種情況下的一些 use case 提案重點(diǎn)從 正反兩面分析了幾種數(shù)據(jù)傳輸提議 包括 Datagrams over DTLS over DCCP over UDP Datagrams over SCTP over DTLS over UDP A new protocol on top of UDP TCP over DTLS over UDP A RTP compatible protocol 重點(diǎn)介紹了 SCTP DTLS UDP 這個協(xié)議?？梢栽趲缀跛胁僮飨到y(tǒng)上得到實(shí)現(xiàn) 認(rèn)為 SCTP 能提供強(qiáng)大的功能 之后 又對 SCTP DTLS UDP 和 DTLS SCTP UDP 做了比較 IETF RTCWeb 草案研究報(bào)告 第頁 4 6 Johnston 提案提案 提案文件 draft johnston rtcweb media privacy 00 涉及內(nèi)容 RTCWEB 媒體隱私與安全 主要內(nèi)容 針對 RTCWEB 媒體安全以及隱私問題 描述了 RTCWEB 需要的安全需求 提 出一個自己主張的 ZRTP 密鑰管理協(xié)議 內(nèi)容概要 Zrtp 是一個完全獨(dú)立的密鑰管理協(xié)議 它最初被設(shè)計(jì)作為 rtp 的擴(kuò)展 但是現(xiàn)在 作為獨(dú)立的協(xié)議運(yùn)行于相同的端口和 ip 地址的 rtp 流 現(xiàn)在 zrtp 同 sip jingle 一起使用 唯一的需求是他們用媒體 rtp 這種靈活性是其他密鑰管理協(xié)議不能 媲美的 它實(shí)現(xiàn)了自己的發(fā)現(xiàn)機(jī)制 它利用 Diffie Hellman 密鑰交換來為 srtp 產(chǎn)生密鑰 Zrtp 避免了使用公共密鑰基礎(chǔ)設(shè)施的需要 而是利用從 SSH 和密鑰 連續(xù)性借鑒的技術(shù) 曾經(jīng)在許多場景媒體不能實(shí)現(xiàn)端到端加密 比如 當(dāng)一個 用戶相信一個服務(wù)器時 對于這些情況 zrtp 開發(fā)了一些機(jī)制用來處理這些情 況 這些都沒有違反基本的安全協(xié)議 或者允許任意的 MITM 實(shí)體在信道內(nèi) 對于 RTCWeb 應(yīng)用 也許有 web 服務(wù)應(yīng)用提供媒體服務(wù) 但是需要信道權(quán)限的 情況 zrtp 能夠支持這些情況 允許用戶明確地對其進(jìn)行授權(quán) 同時不影響 zrtp 的其他優(yōu)點(diǎn) ZRTP 符合 RTCWEB 的媒體安全要求 符合 RTCWEB 的努 力方向 4 7Kaplan 提案提案 提案文件 draft kaplan rtcweb api reqs 01 涉及問題 API Requirements for WebRTC enabled Browsers 主要內(nèi)容 該提案討論了幾項(xiàng)建議方法的優(yōu)點(diǎn)和缺點(diǎn) 關(guān)于什么類型的 API 和 WebRTC 瀏 覽器的結(jié)構(gòu)模型應(yīng)該展現(xiàn)和使用 該提案還提出了一個 API 初始需求列表 內(nèi)容概要 提案討論了在瀏覽器中定義 WebRTC 協(xié)議 包括 session signaling 和 SDP offer answer 的優(yōu)缺點(diǎn) 結(jié)論是 它們不利于應(yīng)用的 webrtc 化 總體上說是 反對這個 web 應(yīng)用模型的 認(rèn)為應(yīng)該依托強(qiáng)大的 JS 庫 將邏輯控制留給 web 開發(fā)人員 因此 這個文件還定義了對于 API 的一些他們認(rèn)為更適合瀏覽器 的初始需求列表 提案文件 draft kaplan rtcweb sip interworking requirements 01 涉及問題 Requirements for Interworking WebRTC with Current SIP Deployments 主要內(nèi)容 This document lists some WebRTC to SIP use cases the WebRTC requirements to support such and the complexity involved in interworking if the requirements cannot be met 內(nèi)容概要 The goal of this document is to summarize the use cases for communicating with deployed SIP devices and domains and capture the requirements necessary to do so without using an Interworking Function or to minimize its cost complexity The impacts or difficulties with various Interworking Function needs are also discussed in order to try to minimize the cost and complexity of using them IETF RTCWeb 草案研究報(bào)告 第頁 4 8 Lennox 提案提案 提案文件 draft lennox rtcweb rtp media type mux 00 涉及問題 通過單一 RTP session 傳輸媒體流的機(jī)制和推薦做法 主要內(nèi)容 該提案描述了如何利用單一 RTP session 來實(shí)現(xiàn)多媒體流的傳輸 內(nèi)容概要 1 利用單一 RTP session 來實(shí)現(xiàn)多媒體流的傳輸?shù)牟襟E 1 Each stream of every media type is a distinct source distinct stream of consecutive packets to be sent to a decoder and is given a distinct synchronization source ID SSRC and has its own distinct timestamp and sequence number space 2 Every media type full media type and subtype e g video h264 or audio pcmu has a distinct payload type value The same payload type value mappings apply across all sources in the session 3 RTP SSRCs initial sequence numbers and initial timestamps are chosen at random independently for each source of each media type 4 RTCP bandwidth is five percent of the total RTP session bandwidth 5 RTP session bandwidth and RTCP bandwidth are divided among all the sources in the session 6 RTCP sender report SR or receiver report RR packets and source description SDES packets are sent periodically for every source in the session 2 向后兼容性 對于 RTCWEB 預(yù)計(jì)要全程使用 ICE 4 9 Nandakumar 提案提案 提案文件 draft nandakumar rtcweb turn uri 00 涉及問題 TURN 協(xié)議 URI 計(jì)劃 主要內(nèi)容 This document is the specification of the syntax and semantics of the Uniform Resource Identifier URI scheme for the Traversal Using Relays around NAT TURN protocol 內(nèi)容概要 URI 計(jì)劃定義包括 URI 語法定義和 URI 語義定義 1 URI 語法 turn URI 的非規(guī)范定義 turn turns turn URI 的規(guī)范定義 IETF RTCWeb 草案研究報(bào)告 第頁 2 URI 語義定義 Turn 協(xié)議支持通過 UDP TCP 和 TLS over TCP 發(fā)送消息 表示 TURN 服務(wù)器 用來識別長期驗(yàn)證的 提案文件 draft nandakumar rtcweb stun uri 00 涉及問題 STUN 協(xié)議 URI 計(jì)劃 主要內(nèi)容 This document is the specification of the syntax and semantics of the Uniform Resource Identifier URI scheme for the Session Traversal Utiliti