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SIP系列课程开讲

2017-09-19 16:06:34   作者:james.zhu   来源:CTI论坛   评论:0  点击:

  根据一些客户的建议,为了让中国客户和通信行业的朋友能够快速掌握VoIP的相关技术内容,本人计划开办一个关于SIP和相关技术的系列讲座。本系列的内容涵盖了十个章节的内容,从传统的PSTN,SIP,传真,云托管,语音问题,IPPBX,SBC,NAT,ICE等等安全问题,和融合通信的基本概念。
  整个讲座课程以文字的形式呈现给用户,同时配有相关的图例。同时,笔者可能需要结合一些开源软交换的具体实例来解释这些功能,例如OpenSIPs, Kamailio,Asterisk等。另外,因个人能力和时间的关系,我们发布的时间可能不是太固定,还有内容的权重可能不一样,内容选题可能有所调整,提前告知大家。但是笔者会按照这个大纲来逐步介绍。
Part 1:SIP 相关基础介绍 SIP – Who Benefits Why SIP? What is SIP? SIP ‘from the RFC’ 3261 New RFCs IETF Working groups Based on HTTP SIP Clients and Servers SIP User Agents Simple Call Session Setup SIP System Architecture The URI - Unique Resource Identifier SIP Addressing SIP Addressing 举例 SIP Servers 和操作 Registration Re-Registration 为什么需要 SIP Proxy servers  Proxy Server ‘State’ types DHCP and SIP SIP Proxy – Trapezoid Model SIP Server – Proxy Mode SIP Server – Re-Direct Mode Location Services SIP Server in Proxy Mode SIP Server in Proxy Redirect Mode Stateful and Stateless Proxies Location Server Location Server – Components Location Server – Information Sources Location Server – Example SIP Client Configuration Configuration scenarios SIP Messaging Request Methods Response Codes SIP Headers INVITE – Example RESPONSE (200 OK) – Example More on Headers Support and Require Headers Timer (Session Times) 100rel (PRACK) Short form ‘compact’ Headers SDP – the Session Description Protocol SDP in a SIP Message 一个SDP 实例 Extending SDP Multiple ‘m’ lines Changing Session Parameters SDP Example - Put a call on Hold SDP Example - Call Hold Trace Call Hold – Old and New Methods Music on Hold example INVITE and reINVITE SIP Mobility SIP Mobility SIP Call Forking - Parallel SIP Call Forking - Sequential Call legs, dialogs and Call IDs Dialog trace example Dialogs and Transactions Branch Ids Call Forward to Voicemail Call Forward - No Answer Replaces header Diversion headers More on Proxies and SIP Routing Stateless Proxy Stateful Proxy More Proxy information VIA and Record Route VIA Details Record-Route Defined Record Route Example Loose and Strict Routing Session Policies MIME MIME Multiple MIME parts SIP and the PSTN SIP and the PSTN SIP to PSTN Detail SIP to PSTN Call Flow SIP Codes and the PSTN SIP and B2BUA B2BUA - Back to Back User Agent B2BUA Example B2BUA Benefits and Features SIP ‘Call Process’ Summary The Call Process  
Part 2:Wireshark 工具
Wireshark What is Wireshark? Download Wireshark Wireshark Introduction Menus, Screens and Views Capturing traffic Profiles Display Filters Capture Filters SIP Packet Analysis SIP ladders and Audio Playback Other Menu options SIP INVITE Analysis Follow a UDP Stream Frame Relationships Colouring Rules RTP Streams View Captures in the ‘Cloud’ What are the codes?
Part 3:SIP/PSTN 介绍
  SIP-T and the PSTN SIP to PSTN Overview SIP to PSTN Call Flow SIP to PSTN Detail PSTN to SIP Call Flow SIP to PSTN Call Failure SIP to PSTN Call trace Early Media Early Media - SIP to PSTN Call Early Offer and Delayed Offer Early Offer / Delayed Offer Gateways Default Gateway? Gateway Location and Routing with TRIP TRIP Examples SIP-T and PSTN Bridging SIP-T and SIP-I SS7, ISDN and SIP ISUP and SIP Messages ISDN User Part (ISUP) to SIP Codes PSTN to PSTN via SIP ISUP Encapsulation ISUP Encapsulation / SDP Addressing Notes SIP and DTMF DTMF - Quick Re-Cap What is DTMF? DTMF Transport methods DTMF ‘Inband’ RFC 2833 ‘Trace’ example RFC 4733 replaces 2833 RFC 4734 SIP INFO 6086 RFC 2833 ‘Trace’ example SIP INFO ‘Trace’ example  
Part 4:SIP/QOS/RTP介绍
What is VoIP or Voice over IP? What is VoIP? What is Voice over IP? VoIP – ‘A Basic Call’ VoIP and TCP / UDP VoIP over the Internet Branch to Branch VoIP Signaling paths Speech paths IP PBX Voice Sampling and Codec Encoding Codecs for Voice Try the Codec Test High Definition (HD) Voice Sound tests Wideband (HD) codecs Opus codec Opus audio examples Codec choices and MOS – Mean Opinion scores Packet Rate / Packets per second The Real Time Protocol or RTP RTP Intro RTP Encapsulation RTP Header Trace Real Time Control Protocol (RTCP) RTCP-XR (Extended Reports) RTP / RTCP and UDP Ports Quality of Service QoS described QoS Issues Measuring Delay Jitter and Packet Loss General VoIP Acceptance Criteria QoS across all Networks 802.1Q – VLANs 802.1Q/P Tagging 802.1P - L2 Classification TOS and DiffServe Layer 3 Classification DSCP with Assured forwarding (AF) Bandwidth decisions Link options – Symmetric DSL (SDSL) Bandwidth (kbps) vs. Packet per Second (pps) Network Behavior Analysis Issues that can affect QoS SIP trunking SIP, SDP and VoIP SIP in the TCP/IP Model SIP and SDP Messages (e.g. Invite and 200OK) SIP and SDP Codec mapping Video over IP What is Video over IP? Streaming Voice and Video – 1 Way Transmission Two-way Conferencing with RTP Codec and Bandwidth Considerations Video bitrate Calculator Setting Video Codecs on Devices Audio and Video in the SDP body Assured SIP Services Assured SIP intro Service Provider Architecture Proxy and Access Router functions Resource-Priority Video ‘example’ Reason Header for Pre-emption Events More Proxy details Multi-Level Pre-emption and Precedence (MLPP)      
Part 5:SIP Security
  Authentication and Authorization SIP Proxy Authentication 401 and 407 Authorization SIP Authorization PROXY Authentication SSL with MD5 Cracked! MD5 v SHA Encryption Why Encrypt SIP? Certificates and HTTPS Certificate Authorities Certificate Example Self-Signed Certificates Format type Securing SIP and VoIP SSL and TLS SIP and TLS TLS Thoughts TLS and SIP in Action SIPS and SIP Addressing Secure RTP (SRTP) Setting SRTP on SIP Devices Secure RTP (SRTP) - Example SRTP and SRTCP sdes and the Crypto attribute Crypto attribute example SRTP Call example ‘showing’ Crypto SRTP with ZRTP RFC 4474 for Caller Identity Caller Identity DTLS/SRTP Ongoing developments for Identity S/MIME and SIP MIME and ISUP SIP Trunking and Security Enhancing SIP Trunk Security Attacks and Responses Types of Attack on a VoIP/SIP Network Responses and Protection Response Identity – A Problem! Rogue SIP Proxy Phishing and SIP exploit More Examples RFC 4475 Try for yourself with ‘example’ software tools NIST Recommendations NIST Recommendations on securing VoIP  
Part 6:防火墙,NAT 和SBC
  Overview Issues to address Firewalls What does a Firewall do? Are Firewalls effective? NAT or Network Address Translation What is NAT? NAT Request NAT Response UDP Hole punching Hairpinning Multiple NATs The NAT Problem Types of NAT Types of NAT NAT – Full Cone NAT – Restricted Cone NAT – Port Restricted Cone NAT – Symmetric The NAPT or (PAT) Problem Problems with NAT, Firewalls and SIP 解决办法 STUN (Session Traversal Utilities for NAT) STUN and rport Problems with ‘Classic’ STUN TURN (Traversal Using Relays around NAT) STUN RFC 5389 Interactive Connectivity Establishment (ICE) ICE ‘In Theory’ Candidate information and other ‘ICE stuff’. ICE ‘In practice’ ICE tags ICE-Lite and Trickle-ICE ICE Client settings More on ICE Universal Plug and Play (UPnP) ‘Near end’ NAT ‘Far end’ NAT GRUU (Globally Routable User Agent) The RTP Problem The Firewall Problem Solving the RTP Problem Symmetric RTP Media Proxy Application Level Gateway SIP Aware Firewalls -呼入 SIP Aware Firewalls - 呼出 Session Border Controllers SBC for the Enterprise and SBC for the ITSP Recommended Session Border Controller features SBCs in Action! SBCs and message manipulation / normalization SIP ‘Refer’ problems SBC ‘Interop’ example SBC Manufacturers - examples From SIP to WebRTC (and back)    
Part 7:SIP 中继介绍和业务要求
SIP Trunks What is a SIP Trunk Alternative to TDM Separate Data and Voice connections Converging the network SIP Trunks and Codecs SIP Trunk Benefits SIP Trunking – In More Depth SIP Trunk Capabilities SIP Trunking Network Examples SIP Peering Peering problems? Least Cost routing (LCR) Disaster Recovery Disaster Recovery ‘Expanded detail’ Disaster Recovery – Last resort? Number Consolidation Virtual Presences Trunking Variations Single Site, No ‘Forklift’ Single Site, TDM PBX Single Site, Converged Converged – SIP/IP PBX Multiple Site, ‘Converged’ Multiple Site, ‘Converged’ + central SBC Multiple Site, ‘Converged’ + Multiple SBCs Media Gateways SIP PBX to Non-SIP PBX SIP PBX to Non-SIP PBX, Call Flow SIP Trunk Performance Connection types The ADSL issue Codecs, Voice and Data Symmetric DSL (SDSL) Bandwidth Calculator Testing your link ADSL Developments Fibre Options SIP Trunking, MPLS and SD-WAN MPLS, basic explanation MPLS Label format MPLS in a MAC frame MPLS example network MPLS benefits Your own private WAN but ‘Not the only client’ Separate MPLS networks VPLS explained WAN Optimization, Hybrids and SD-WAN Software Defined WANs explained Security and SIP Trunking SIP Trunk Security - Overview Session Border Controllers More on SBCs The ‘corporate’ SBC SIP REFER issues Setting up a SIP Trunk Add a VoIP Provider Provider SIP Servers Authentication Add a Dialling Rule Trunk setup complete Call out Trace Comparing SIP packets from two ITSP providers Skype for Business and SIP trunks ‘Optional’ Lab exercises Skype for Business ‘Network Environment’ Topology Builder Control Panel Management Shell and basic commands Installing Skype for Business Client Making Calls Using Wireshark to monitor calls from a Skype network environment to the PSTN across a SIP trunk Some PBX Requirements Enterprise PSTN Identities P-Preferred and P-Asserted Call Progress Tones Troubleshooting and Interops SIP Trunks and Common Problems Choosing an ITSP Understanding ITSP Offerings 'Sticking points’? What you may need in the future SIP trunk ‘connectivity’ Things to watch out for when connecting to your ITSP ‘Finding’ an ITSP SIP trunking Checklist for ITSP evaluation Working together SIP trunk connectivity items ‘from the field’    
Part 8:SIP 和 Fax over IP
Faxing Basics Faxing background T.30 Fax signaling Associated tones and protocols The ITU and TIA standards Fax over IP Fax over IP benefits From the old to the new Intro to FoIP FoIP and SIP trunks Protocol conversions Fax Protocols G.711 Pass-through T.37 Store and Forward T.38 Relay Where does SIP fit in? UDPTL Protocol options for the future FoIP in action SIP in FoIP – Call Flow SIP INVITE INVITE for T.38 The INVITE SDP body Wireshark FoIP example SIP T.38 Call flows – IETF draft document Bandwidth T.38 and G.711 network traffic Troubleshooting The basics More complex issues to watch out for Ongoing Efforts RFC 6913 and sip.fax tag Use DTMF events instead?
Part 9:SIP和UC 融合通信介绍
Communication Breakdown Playing Voicemail tag Can’t find people Available but not Available..! More Examples of communication problems IM Clients IM Client Examples and Features More in IM Clients The Background Stuff The IMPP working group IMPP and CPP More IMPP work SIMPLE How it all works Presentity A Basic SIP subscription Multiple Presence States Presence and P2P A Presence Network Getting inside the SIP packets Presentity and more! A Basic SIP Subscription Multiple Presence States Presence and P2P A Presence Network Get inside the SIP packets The Packet Structure PIDF Message Body XML Tuples Example Presence doc with Tuples (using a Mobile Phone) The METHODS in Action PUBLISH SUBSCRIBE NOTIFY MESSAGE is-composing Rich Presence 2 Places at the same time ‘Presence’ Federations What is Federation? Multiple Presence sources Super-Aggregation Inter-Domain Federation Conferencing What SIP does in Conferencing INITIATE a conference JOIN a conference LEAVE / EXIT a conference INVITE other participants REFER conference server to invite or others to join EXPEL participants CONFIGURE the media stream CONTROL a conference Why SIP? Centralized conferencing Centralized Signaling Centralized Mixing (optional) Centralized Authentication B2BUA (Discussed in core module) Conference Components The Focus More than one Focus Creating a Conference Creating a Conference: Details Adding a participant Adding a participant: Details Alternative INVITE with REFER Unified Communications What’s all the fuss? Unified Confusion What is Unified Communications? From UC to UCaaS Components involved What should UC do? 21st Century Dial tone The Unified inbox Unified aware applications Find me – Follow me Device awareness Unified Comms for Business Humans and UC Migrating to UCaaS UCasS, SIP and the WAN    
Part 10:SIP,云托管,LTE,IMS 介绍
Hosted SIP What Hosted SIP service is Hosted functions and features Example Network including ‘failover’ ‘Hosted’ clients in action Why Hosted – Benefits and things to consider Why on-site PBX – Benefits and things to consider Auto Provisioning Auto Provisioning Example Boot Server Client Config Client boot sequence Client config download RFC 6011 Benefits of Hosted SIP Service Benefits of Onsite PBX and SIP trunks SIP, LTE, the IMS and VoLTE Network Overview RAN, eNodeB, EPC, IP Core and 3GPP 4G, LTE, LTE Advanced, WiMAX2 The RAN and EPC Default Bearer Setup Introduction to the Servers and Functions in the IMS CSCF S-CSCF P-CSCF I-CSCF Home Subscriber Server HSS Application Server TAS PSCF DNS and ENUM Device Registration (with SIP) SIP Registration packet example SIP in the IMS – Call Flow explained Introduction to VoLTE and the threat of OTT services Making VoLTE work SIP Preconditions in Action With Codec examples within SDP SIP Call flow for VoLTE Quality settings ‘recap’ VoLTE media flow More on VoLTE The IMS Layers architecture Application IMS / Session Control Access and Transport 3GPP Multiple access devices RCS and OTT Who provides IMS solutions? IPX and Peering for Security, QoS and SLAs GSMA and IR.92 HD Voice News SIP and Fax over IP G.711 Pass-through T.37 Store and Forward T.38 Relay UDPTL Protocol options for the future FoIP in action SIP in FoIP – Call Flow SIP INVITE INVITE for T.38 The INVITE SDP body Wireshark FoIP example SIP T.38 Call flows – IETF draft document Bandwidth T.38 and G.711 network traffic Troubleshooting The basics More complex issues to watch out for Ongoing Efforts RFC 6913 and sip.fax tag Use DTMF events instead
  以上是我们计划开讲的所有基本内容,希望给大家分享一些真正有价值的SIP相关技术资料,和大家一起进步!
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