这章主要涉及internet telephony。编码标准，RTP,SIP, 和一个example sdterisk
1. Why need Internet telephony?(经济因素，统一的硬件，支持移动设备，增加services)
- Economical perspective:
1. reduce investment costs for network hardware
2. same infrastructure for all classes of data/reduced amount of protocols
3. efficient coding possible
- enhanced functionality/flexibility
1. supports mobility of users/devices
2.can be easily extended by further services
2. Transformation to PDUs
- Bit stream: stream of audio data generated continuously by telephony application
- Logical Data Unit(LDU): Media dependent assembly of packets, e.g. blocks of individual samples
- Protocol Data Unit(PDU): packet of a communication protocol consisting of header and payload part.
3. Audio coding :
- continuous sinal and frequency range : 300-3400hz. --> low-pass filter: smoothing and removing of high-frequency noise-->sampling:sample rate >2*frequency--->coding:sample-based:single samples are coded or segment based : segnment of 10-30ms are coded.
这部分算是通信的内容，主要涉及到了采样定理和信号特性。
4.comparison of codecs
对基本coding标准进行介绍： compression 越低， delay越小；data rate是一个重要的参考
5. Protocol:
5.1 结构 从顶向下 coding/decoding / transport layer/ network layer.
显然我们需要使用ip 在network layer，用t c p/ udp 在transport layer以保证现有的设备可以被使用，但是我们需要在tcp/udp之上增加一个协议来规范audio communication.
dedicated protocols and mechanisms needed for:
- signaling and session initiation / termination
- Transport of audio data
- Addressing of participants
- Interoperability with further communication/ telephony approaches
5.2 ENUM
- ENUM DNS offers mapping of telephony numbers to URIs using NAPTR(Naming Authority Pointer) resource records
- 数字反向，IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:alice@a.de!"
- 2 types: Carrier-ENUM/ USer-ENUM:
carrier-ENUM: send request containing the queried user telephone number to SIP-Server, SIP server transfer the telephone number to NAPTR resource records and forward it to the ENUM server then Carrier-ENUM returns the URIs;SIP server find the SIP-server which is responsible for the queried user. And this server forward request to queried user.
user-ENUM: users send requests to ENUM server directly using NAPTR records and get the URI of the sip-server which is repsonsile for the queried users. Then users can forward requests to SIP-Server directly and try to build up communication between queried Users directly. when quried user get permission information from sip-server, the communciation has been built.
5.3 some terms including : LDAP: Lightweight Directory Access Protocol --located user server; TRIP: telephony routing over IP - exchange of routing information; MG: media gateway; MGC: media gateway controller;Megaco: Media Gateway control protocol; MGCP- Media Gateway control protocol;PSTN - Public switched Telephone Network.
5.4 RTP: real-time transport protocol
- Real-time layer is located in transport layer and application layer.
- Used in huge amount of audio and video communication systems
- Functionality: Identification of media source (source ID); Packet ordering(source ID); Synchronisation(via timestamp):removal of Jitter, synchronisation of audio and video; Dynamic flow control(using RTCP)
- Real-time transport based on UDP
- Audio packets contain 10-50 ms of audio data
- RTP adds at least 12 Byte header for each packet
- Used Codecs are specified by an ID
首先RTP位于传输层和应用层中间，然后主要功能是进行多媒体包源标示，包顺序，同步。分别用identifier of media source， sequence number, timestamp. 动态流控制用RTCP。 基于UDP，是segment-based coding.
5.5 RTP header(minimum of 12 Bytes)
5.6 RTCP - RTP control protocol
- RTCP = support protocol for RTP
- Used for: feedback on QoS parameters(packet losses and RTTs)--> used for flow control and error correction; identification of participants of a RTP session
- RTP specification defines five RTCP packet types:
1. SR: sender report : transmission and reception statistics from active senders
2.RR:Receiver report: Reception statistics from receivers; Contains e.g. last sequence number, interarrival jitter, commulative number of lost packets.
3. SDES: source description items: Information about the source, e.g. name and e-mail.
4.BYE: Indification of end of participation
5.APP: application-specific functions
总之，RTCP是RTP的支撑协议，主要用于feedback - error correction , flow control; identification of participants of a RTP session； 4种类型包：SR, RR, SDES, BYE, APP
5.5 SIP
is used for creating, modifying, and terminating sessions that works independently of underlying transport protocols and without dependency on the type of session that is being established.
- session = "exchange of data between an association of participants"
- challenges: 1. mobility of participants/2.availability via several names/3.communication via different media.
- functionality of sip: 1.localization of participants/ 2.session establishment/ 3.presence management/ 4.negotiation of session parameters such as used audio/video codecs
- user agent contains a user agent server and a user agent client part
- userclient is generating a request based on some external event and processes responses
- userserver receives requests and generates response
- sip is a transcational protocol:
1. one transcation consists of a single request and one or several response
2. requests are sent by client transactions
3. responses are sent by server transactions
- messages:
1. message structure is similar to HTTP 1.0/1.1:
1.1 Text messages
1.2 Header contains
1.3method/URI/SIP version/Key-value pairs such as "to" or "from"
2. Body depends on SIP method and contians e.g. SDP data
3. Responses contain status/success/error code plus associated message
- Requests:
1. six important request types:
1.1Register:1. manages bindings between a SIP or SIP URI and an IP address/2.Used to add, remove or query for bindings
2. INVITE: most important request: establishes session between user agents and is used to negotiate session parameters
3. ACK: confirms session establishment
4. BYE: terminates a request
5. OPTIONS: used to request features of communication partner
- Response:1.1xx provisional responses/2.success3/redirect/4.bad request/5.server error/6.global error
5.6 SDP exmaple
- session description protocol defined in RFC 4566
- Used to describe multimedia session during session establishment
- Used by SIP and further protocols such as RTSP
- text based format
- v= protocol version
- o = initiator of session
- c = information about connection
- m =media name
- a =media attributes
- component:
User Agent: client/server application running on end user devices
Proxy server: forwards requests to further SIP domains
redirect server : redirects requests to new location of UA
registrar: maps sip names to address
- SIP uniform resource identifier
1.URI scheme alternative: "sip" for secure connections
local user name, e.g. "alice"
associated domain name, e.g. "a.de"
sip:user@domain
2. goal: unique identification of users and services independent from current location, e.g.:sip:alice@a.de
3. alternatively,"tel" URLs for telephony calls may be used
4. focus of SIP is directed to mobility:
- user/service mobility: interaction with user/services is independent from location of user/service
- device mobility: changine ip addresses/ temporary unavailablity are supported
- session mobility: sip session can be transfered to another device
5.location of users via SIP URI
- DNS registeration:
1.SRV(service) resource records of the DNS(domain name system ) can be used to identify services associated to a specific domain
2. SRV resource record contains the service name, the transport protocol and the associated domain name plus a port number, a priority and weight information and the name of the host providing the service
3. Usage in SIP:
address of sip proxy: address of SIP proxies may point to different domain; renders several physical SIP proxies per logical proxy possible: sip:a.de->sip1.a.de,sip2.a.de
one physical sip proxy for several logical proxies
sip:a.de, sip:d.de -? sip.iptel.org
- SIP registeration
1. registration is done using the REGISTER request
2. usually requires authentication
3.registrar manages addresses and collaborates with Proxy in same domain
4. New registration necessary after a specific timespan
- address resolution via dns:
1. alice@a.de ->DNS query(SRV, sip.tcp.b.de)->DNS(determinde SIP proxy of b.de via SRV query)
2. proxy.b.de
3. alice@a.de -> (a record, proxy.b.de) -> DNS(determine IP of proxy)
4. a record:24.2.2.3
5. invite -> proxy(determine ip of bob via location service )-->invite bob@b.de
6. SIP cases:scenario（剧本）-based discussion of SIP
6.1 peer-to-peer
both uAs are located in different subnets; access via ip addresses possible;UAs establish a session directly; IP is determined via Redirect server; Direct RTP session for media transfer
6.2proxy
one procy for each domain; registration is only done at local register;ip address is determined within destination domain;direct RTP session(IP and port of the reply to INVITE);recoding of routes - response on the same route:follow-up request directly or follow -up on recorded route.
6.3 SIP -> telephony network
call initiated by sip ua; telephon number is provided as "tel" uri; sip proxy determines via location server sip URI of responsible gateway; SIP gateways controls media gateway; RTP session is established between UA and MG; Transformation to regular telephony connection between MG and phone; communication initiated by conventional telephone network; sip ua has assigned phone number via ENUM to SIP URI; SIP address is determined using a service control point(SCP); Gateway establishes connection to sip ua
