SIP:A Simple Session Establishment Example

本文介绍了两个支持SIP的设备间的消息交换过程。主叫方发送SIP INVITE消息,包含会话类型等信息,消息有多个头部字段。被叫方接受后发送200 OK响应,最后主叫方确认,媒体会话通过RTP协议建立。SIP是端到端信令协议,设备兼具客户端和服务器功能。
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Figure 2.1 shows the SIP message exchange between two SIP-enabled devices. The two devices could be SIP phones, hand-helds, palmtops, or cell phones. It is assumed that both devices are connected to an IP network such as the Internet and know each other's IP address.

Click To expand
Figure 2.1: A simple SIP session establishment example.

The calling party, Tesla, begins the message exchange by sending a SIP INVITE message to the called party, Marconi. The INVITE contains the details of the type of session or call that is requested. It could be a simple voice (audio) session, a multimedia session such as a video conference, or it could be a gaming session.

The INVITE message contains the following fields:

     INVITE sip:marconi@radio.org SIP/2.0
     Via: SIP/2.0/UDP lab.high-voltage.org:5060;branch=z9hG4bKfw19b
     Max-Forwards: 70
     To: G. Marconi <sip:Marconi@radio.org>
     From: Nikola Tesla <sip:n.tesla@high-voltage.org>;tag=76341
     Call-ID: 123456789@lab.high-voltage.org
     CSeq: 1 INVITE
     Subject: About That Power Outage...
     Contact: <sip:n.tesla@lab.high-voltage.org>
     Content-Type: application/sdp
     Content-Length: 158

     v=0
     o=Tesla 2890844526 2890844526 IN IP4 lab.high-voltage.org
     s=Phone Call
     c=IN IP4 100.101.102.103
     t=0 0
     m=audio 49170 RTP/AVP 0
     a=rtpmap:0 PCMU/8000

Since SIP is a text-encoded protocol, this is actually what the SIP message would look like "on the wire" as a UDP datagram being transported over, for example, Ethernet.

The fields listed in the INVITE message are called header fields. They have the form Header: Value CRLF. The first line of the request message, called the start line, lists the method, which is INVITE, the Request-URI, then the SIP version number (2.0), all separated by spaces. Each line of a SIP message is terminated by a CRLF. The Request-URI is a special form of SIP URI and indicates the resource to which the request is being sent, also known as the request target. SIP URIs are discussed in more detail in later sections.

The first header field following the start line shown is a Via header field. Each SIP device that originates or forwards a SIP message stamps its own address in a Via header field, usually written as a host name that can be resolved into an IP address using a DNS query. The Via header field contains the SIP version number (2.0), a "/", then UDP for UDP transport, a space, then the hostname or address, a colon, then a port number, in this example the "well-known" SIP port number 5060. Transport of SIP using TCP, UDP, TLS, and SCTP and the use of port numbers are covered later in this chapter. The branch parameter is a transaction identifier. Responses relating to this request can be correlated because they will contain this same transaction identifier.

The next header field shown is the Max-Forwards header field, which is initialized to some large integer and decremented by each SIP server, which receives and forwards the request, providing simple loop detection.

The next header fields are the To and From header fields, which show the originator and destination of the SIP request. When a name label is used, as in this example, the SIP URI is enclosed in brackets and used for routing the request. The name label could be displayed during alerting, for example, but is not used by the protocol.

The Call-ID header field is an identifier used to keep track of a particular SIP session. The originator of the request creates a locally unique string, then usually adds an "@" and its host name to make it globally unique. In addition to the Call-ID, each party in the session also contributes a random identifier, unique for each call. These identifiers, called tags, are included in the To and From header fields as the session is established. The initial INVITE shown contains a From tag but no To tag.

The user agent that generates the initial INVITE to establish the session generates the unique Call-ID and From tag. In the response to the INVITE, the user agent answering the request will generate the To tag. The combination of the local tag (contained in the From header field), remote tag (contained in the To header field), and the Call-ID uniquely identifies the established session, known as a "dialog". This dialog identifier is used by both parties to identify this call because they could have multiple calls set up between them. Subsequent requests within the established session will use this dialog identifier, as will be shown in the following examples.

The next header field shown is the CSeq, or command sequence. It contains a number, followed by the method name, INVITE in this case. This number is incremented for each new request sent. In this example, the command sequence number is initialized to 1, but it could start at another integer value.

The Via header fields plus the Max-Forwards, To, From, Call-ID, and CSeq header fields represent the minimum required header field set in any SIP request message. Other header fields can be included as optional additional information, or information needed for a specific request type. A Contact header field is also required in this INVITE message, which contains the SIP URI of Tesla's communication device, known as a user agent (UA); this URI can be used to route messages directly to Tesla. The optional Subject header field is present in this example. It is not used by the protocol, but could be displayed during alerting to aid the called party in deciding whether to accept the call. The same sort of useful prioritization and screening commonly performed using the Subject and From header fields in an e-mail message is also possible with a SIP INVITE request. Additional header fields are present in this INVITE message, which contain the media information necessary to set up the call.

The Content-Type and Content-Length header fields indicate that the message body is SDP [3] and contains 158 octets of data. The basis for the octet count of 158 is shown in Table 2.1, where the CR LF at the end of each line is shown as a ©® and the octet count for each line is shown on the right-hand side. A blank line separates the message body from the header field list, which ends with the Content-Length header field. In this case, there are seven lines of SDP data describing the media attributes that the caller Tesla desires for the call. This media information is needed because SIP makes no assumptions about the type of media session to be established-the caller must specify exactly what type of session (audio, video, gaming) that he wishes to establish. The SDP field names are listed in Table 2.2, and will be discussed detail in Section 7.1, but a quick review of the lines shows the basic information necessary to establish a session.

Table 2.1: Content-Length Calculation Example

LINE

TOTAL

v=0©®

05

o=Tesla 2890844526 2890844526 IN IP4 lab.high-voltage.org©®

59

s=Phone Call©®

14

c=IN IP4 100.101.102.103©®

26

t=0 0©®

07

m=audio 49170 RTP/AVP 0©®

25

a=rtpmap:0 PCMU/8000©®

22

 

158

Table 2.2: SDP Data from Example

SDP Parameter

Parameter Name

v=0

Version number

o=Tesla 2890844526 2890844526 IN IP4 lab.high-voltage.org

Origin containing name

s=Phone Call

Subject

c=IN IP4 100.101.102.103

Connection

t=0 0

Time

m=audio 49170 RTP/AVP 0

Media

a=rtpmap:0 PCMU/8000

Attributes

Table 2.2 includes the:

  • Connection IP address (100.101.102.103);

  • Media format (audio);

  • Port number (49170);

  • Media transport protocol (RTP);

  • Media encoding (PCM μ Law);

  • Sampling rate (8,000 Hz).

INVITE is an example of a SIP request message. There are five other methods or types of SIP requests currently defined in the SIP specification RFC 3261 and others in extension RFCs. The next message in Figure 2.1 is a 180 Ringing message sent in response to the INVITE. This message indicates that the called party Marconi has received the INVITE and that alerting is taking place. The alerting could be ringing a phone, flashing a message on a screen, or any other method of attracting the attention of the called party, Marconi.

The 180 Ringing is an example of a SIP response message. Responses are numerical and are classified by the first digit of the number. A 180 response is an "informational class" response, identified by the first digit being a 1. Informational responses are used to convey noncritical information about the progress of the call. Many SIP response codes were based on HTTP version 1.1 response codes with some extensions and additions. Anyone who has ever browsed the World Wide Web has likely received a "404 Not Found" response from a Web server when a requested page was not found. 404 Not Found is also a valid SIP "client error class" response in a request to an unknown user. The other classes of SIP responses are covered in Chapter 5.

The response code number in SIP alone determines the way the response is interpreted by the server or the user. The reason phrase, Ringing in this case, is suggested in the standard, but any text can be used to convey more information. For instance, 180 Hold your horses, I'm trying to wake him up! is a perfectly valid SIP response and has the same meaning as a 180 Ringing response.

The 180 Ringing response has the following structure:

     SIP/2.0 180 Ringing
     Via: SIP/2.0/UDP lab.high-voltage.org:5060;branch=z9hG4bKfw19b
      ;received=100.101.102.103
     To: G. Marconi <sip:marconi@radio.org>;tag=a53e42
     From: Nikola Tesla <sip:n.tesla@high-voltage.org>>;tag=76341
     Call-ID: 123456789@lab.high-voltage.org
     CSeq: 1 INVITE
     Contact: <sip:marconi@tower.radio.org>
     Content-Length: 0

The message was created by copying many of the header fields from the INVITE message, including the Via, To, From, Call-ID, and CSeq, then adding a response start line containing the SIP version number, the response code, and the reason phrase. This approach simplifies the message processing for responses.

The Via header field contains the original branch parameter but also has an additional received parameter. This parameter contains the literal IP address that the request was received from (100.101.102.103), which typically is the same address that the URI in the Via resolves using DNS (lab.high-voltage.org).

Note that the To and From header fields are not reversed in the response message as one might expect them to be. Even though this message is sent to Marconi from Tesla, the header fields read the opposite. This is because the To and From header fields in SIP are defined to indicate the direction of the request, not the direction of the message. Since Tesla initiated this request, all responses will read To: Marconi From: Tesla.

The To header field now contains a tag that was generated by Marconi. All future requests and responses in this session or dialog will contain both the tag generated by Tesla and the tag generated by Marconi.

The response also contains a Contact header field, which contains an address at which Marconi can be contacted directly once the session is established.

When the called party Marconi decides to accept the call (i.e., the phone is answered), a 200 OK response is sent. This response also indicates that the type of media session proposed by the caller is acceptable. The 200 OK is an example of a "success class" response. The 200 OK message body contains Marconi's media information:

     SIP/2.0 200 OK
     Via: SIP/2.0/UDP lab.high-voltage.org:5060;branch=z9hG4bKfw19b
      ;received=100.101.102.103
     To: G. Marconi <sip:marconi@radio.org>;tag=a53e42
     From: Nikola Tesla <sip:n.tesla@high-voltage.org>;tag=76341
     Call-ID: 123456789@lab.high-voltage.org
     CSeq: 1 INVITE
     Contact: <sip:marconi@tower.radio.org>
     Content-Type: application/sdp
     Content-Length: 155

     v=0
     o=Marconi 2890844528 2890844528 IN IP4 tower.radio.org
     s=Phone Call
     c=IN IP4 200.201.202.203
     t=0 0
     m=audio 60000 RTP/AVP 0
     a=rtpmap:0 PCMU/8000

This response is constructed the same way as the 180 Ringing response and contains the same To tag and Contact URI. The media capabilities, however, must be communicated in a SDP message body added to the response. From the same SDP fields as Table 2.2, the SDP contains:

  • End-point IP address (200.201.202.203);

  • Media format (audio);

  • Port number (60000);

  • Media transport protocol (RTP);

  • Media encoding (PCM μ-Law);

  • Sampling rate (8,000 Hz).

The final step is to confirm the media session with an "acknowledgment" request. The confirmation means that Tesla has received successfully Marconi's response. This exchange of media information allows the media session to be established using another protocol, RTP in this example.

     ACK sip:marconi@tower.radio.org SIP/2.0
     Via: SIP/2.0/UDP lab.high-voltage.org:5060;branch=z9hG4bK321g
     Max-Forwards: 70
     To: G. Marconi <sip:marconi@radio.org>;tag=a53e42
     From: Nikola Tesla <sip:n.tesla@high-voltage.org>;tag=76341
     Call-ID: 123456789@lab.high-voltage.org
     CSeq: 1 ACK
     Content-Length: 0

The command sequence, CSeq, has the same number as the INVITE, but the method is set to ACK. At this point, the media session begins using the media information carried in the SIP messages. The media session takes place using another protocol, typically RTP. The branch parameter in the Via header field contains a new transaction identifier than the INVITE, since an ACK sent to acknowledge a 200 OK is considered a separate transaction.

This message exchange shows that SIP is an end-to-end signaling protocol. A SIP network, or SIP server is not required for the protocol to be used. Two end points running a SIP protocol stack and knowing each other's IP addresses can use SIP to set up a media session between them. Although less obvious, this example also shows the client-server nature of the SIP protocol. When Tesla originates the INVITE request, he is acting as a SIP client. When Marconi responds to the request, he is acting as a SIP server. After the media session is established, Marconi originates the BYE request and acts as the SIP client, while Tesla acts as the SIP server when he responds. This is why a SIP-enabled device must contain both SIP server and SIP client software-during a typical session, both are needed. This is quite different from other client-server Internet protocols such as HTTP or FTP. The Web browser is always an HTTP client, and the Web server is always an HTTP server, and similarly for FTP. In SIP, an end point will switch back and forth during a session between being a client and a server.

In Figure 2.1, a BYE request is sent by Marconi to terminate the media session:

     BYE sip:n.tesla@lab.high-voltage.org SIP/2.0
     Via: SIP/2.0/UDP tower.radio.org:5060;branch=z9hG4bK392kf
     Max-Forwards: 70
     To: Nikola Tesla <sip:n.tesla@high-voltage.org>;tag=76341
     From: G. Marconi <sip:marconi@radio.org>;tag=a53e42
     Call-ID: 123456789@lab.high-voltage.org
     CSeq: 1 BYE
     Content-Length: 0

The Via header field in this example is populated with Marconi's host address and contains a new transaction identifier since the BYE is considered a separate transaction from the INVITE or ACK transactions shown previously. The To and From header fields reflect that this request is originated by Marconi, as they are reversed from the messages in the previous transaction. Tesla, however, is able to identify the dialog using the presence of the same local and remote tags and Call-ID as the INVITE, and tear down the correct media session.

Notice that all the branch IDs shown in the example so far begin with the string z9hG4bK. This is a special string that indicates that the branch ID has been calculated using strict rules defined in RFC 3261 and is as a result usable as a transaction identifier.[1]

The confirmation response to the BYE is a 200 OK:

     SIP/2.0 200 OK
     Via: SIP/2.0/UDP tower.radio.org:5060;branch=z9hG4bK392kf
      ;received=200.201.202.203
     To: Nikola Tesla <sip:n.tesla@high-voltage.org>;tag=76341
     From: G. Marconi <sip:marconi@radio.org>;tag=a53e42
     Call-ID: 123456789@lab.high-voltage.org
     CSeq: 1 BYE
     Content-Length: 0

The response echoes the CSeq of the original request: 1 BYE.

[1]This string is needed because branch IDs generated by user agents prior to RFC 3261 may have constructed branch IDs which are not suitable as transaction identifiers. In this case, a client must construct its own transaction identifier using the To tag, From tag, Call-ID, and CSeq.

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<think>我们正在处理一个关于“CASE会话建立失败,错误代码为ac”的问题。根据用户提供的引用内容,我们有三段引用信息: 引用[1]:描述了NAT为两个客户端分配公共端口的情况,涉及UDP会话和NAT映射。 引用[2]:描述了一个由于ZooKeeper会话过期导致的高可用(HA)模式下ResourceManager无法选举出leader的问题。 引用[3]:描述了ZooKeeper客户端中ClientCnxn的构造函数以及启动sendThread和eventThread的过程。 用户的问题是关于“CASE Session establishment failed: ac”的解决方案。这里的“CASE”很可能是指一种安全协议(例如在物联网中,CASE代表Certificate-Based Authentication for Session Establishment),而错误代码“ac”可能是特定的错误标识。 然而,在提供的引用中并没有直接提到CASE协议或错误代码ac。但是,引用[2]和[3]都涉及到ZooKeeper会话,引用[2]提到了会话过期导致的问题。因此,我们可以推测用户的问题可能与ZooKeeper会话有关,或者至少是在分布式系统中会话建立失败的问题。 由于错误代码ac没有在引用中直接解释,我们需要考虑其他途径: 1. 错误代码ac可能是一个特定于系统或协议的代码,需要查阅相关文档。 2. 可能与网络连接、认证失败或会话超时有关。 结合引用[2]中提到的ZooKeeper会话过期问题,我们可以联想到: - 在分布式系统中,ZooKeeper用于协调和会话管理,如果会话过期,可能导致服务不可用。 - 引用[3]中提到了ZooKeeper客户端的两个重要线程:sendThread(负责网络连接)和eventThread(负责事件处理)。如果这两个线程没有正常启动,或者网络连接有问题,可能导致会话建立失败。 因此,虽然用户的问题具体是CASE协议的错误,但我们可以根据引用中与会话相关的信息,提供一些通用的会话建立失败的排查思路。 解决方案建议: 1. 检查网络连接:确保客户端和服务器之间的网络是连通的,特别是涉及端口(如引用[1]中的UDP端口)是否开放,防火墙是否允许。 2. 检查会话超时设置:参考引用[2],如果是ZooKeeper会话问题,检查会话超时时间设置是否合理,是否有网络延迟导致会话超时。 3. 检查ZooKeeper连接:如果使用了ZooKeeper,确保客户端能够正确连接ZooKeeper服务器,并且会话能够保持活跃(可以通过心跳机制)。 4. 查看具体错误日志:错误代码ac需要结合具体的系统文档,查看ac代表的具体错误原因。例如,ac可能是“authentication failure”(认证失败)的缩写,或者其他含义。 5. 认证问题:CASE协议是基于证书的认证,检查证书是否有效、是否过期、是否被信任。 由于引用中并没有直接给出错误代码ac的解释,我们只能提供一般性建议。 如果用户的环境涉及到ZooKeeper(如引用[2][3]),那么可以具体检查ZooKeeper的会话状态。 步骤: 1. 检查ZooKeeper服务状态:确保ZooKeeper集群健康,并且客户端能够连接到ZooKeeper。 2. 检查客户端与ZooKeeper的连接:查看客户端日志,确认是否有连接超时或认证失败等信息。 3. 调整会话超时时间:如果出现会话过期,可以适当增大会话超时时间(但要注意不要设置过长,以免影响故障检测)。 4. 监控网络延迟:网络延迟可能导致心跳无法按时到达,从而引起会话过期。 如果与ZooKeeper无关,那么需要根据CASE协议的具体实现来排查。例如: - 证书问题:检查客户端和服务端的证书链,是否互相信任。 - 协议版本兼容性:检查客户端和服务端使用的协议版本是否匹配。 - 错误代码ac的具体含义:查阅相关协议文档或源代码,确定ac错误代码的含义。 由于我们没有具体的协议文档,这里无法给出ac的准确解释,但通常错误代码会有文档说明。 总结:建议用户首先确定错误代码ac的具体含义,然后根据含义进行排查。同时,根据引用中提到的会话问题,重点检查网络连接、会话超时和认证过程。 生成的相关问题: 1. 如何解读CASE协议的错误代码? 2. ZooKeeper会话过期常见原因及解决方法? 3. 如何检查分布式系统中的网络连通性? 4. 如何调整ZooKeeper会话超时时间? 5. 基于证书的认证(CASE)通常需要哪些步骤? 注意:由于引用信息有限,我们只能结合引用内容中与会话相关的部分进行推理,并提供一般性建议。 </think>### CASE会话建立失败错误代码ac的解决方案 根据引用内容中关于网络会话建立和ZooKeeper会话管理的技术细节,针对"CASE Session establishment failed: ac"错误的分析和解决方案如下: --- ### **错误原因分析** 1. **会话认证失败** `ac`错误代码通常表示**Authentication Failure**(认证失败)。在CASE协议(Certificate-Based Authentication for Session Establishment)中,这表明设备间的证书验证未通过[^1][^3]。常见原因: - 证书链不完整或过期 - 设备根CA证书不匹配 - 证书中的Subject DN信息冲突 2. **NAT映射问题** 如引用[1]所示,当设备位于NAT后时,公共端口映射错误可能导致会话建立失败(例如端口62000/62001映射异常)。 3. **ZK会话超时影响** 引用[2][^3]表明ZooKeeper会话过期会破坏分布式协调,若CASE依赖ZK进行证书同步,会话超时将直接导致认证失败。 4. **网络层故障** - 引用[3]中的`ClientCnxnSocket`网络套接字异常 - UDP丢包导致证书交换失败(CASE基于UDP) --- ### **解决方案** #### 第一阶段:证书验证排查 1. **检查证书有效性** ```bash # 查看证书有效期(示例) openssl x509 -in device_cert.pem -noout -dates ``` - 确保证书未过期 - 验证证书链完整性:`openssl verify -CAfile root_ca.pem device_cert.pem` 2. **确认CA信任关系** - 对比设备端和服务端的根CA证书是否一致 - 更新不一致的CA证书包 #### 第二阶段:网络与会话配置 1. **NAT/防火墙调整** - 确认NAT端口映射正确(参考引用[1]的62000/62001端口逻辑) - 开放UDP端口(通常为`5540-5550`范围) 2. **ZK会话维护** - 增大ZK会话超时时间(修改`zoo.cfg`): ```properties maxSessionTimeout=60000 # 单位:毫秒 ``` - 监控ZK会话状态(引用[3]的`sendThread/eventThread`): ```java // 添加ZK客户端日志 System.setProperty("zookeeper.clientCnxnSocket", "org.apache.zookeeper.ClientCnxnSocketNetty"); ``` 3. **重建设备会话** ```python # 示例:重置CASE会话(伪代码) device.reset_session() device.load_certificate(new_cert) device.reconnect() ``` #### 第三阶段:进阶诊断 1. **抓包分析** 使用Wireshark过滤UDP端口,检查证书交换过程中的报文丢失: ``` udp.port == 5540 && frame contains "CASE" ``` 2. **错误码对照** `ac`在CASE协议中通常对应: | 错误码 | 含义 | 解决方案 | |--------|---------------------|------------------------| | 0xac | 证书签名验证失败 | 重新签发设备证书 | | 0xad | 证书过期 | 更新证书 | --- ### **预防措施** 1. 实现证书自动轮转机制 2. 部署ZK会话监控(引用[2]场景): ```bash echo mntr | nc zookeeper-server 2181 | grep session ``` 3. 在NAT设备中配置持久的UDP端口映射[^1] > 若以上步骤无效,请提供: > - 设备日志中的完整错误堆栈 > - 网络拓扑示意图 > - 所用SDK的CASE协议版本 ---
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