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TWI506998B - Traversal method for icmp-sensitive nat - Google Patents

Traversal method for icmp-sensitive nat Download PDF

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TWI506998B
TWI506998B TW102104700A TW102104700A TWI506998B TW I506998 B TWI506998 B TW I506998B TW 102104700 A TW102104700 A TW 102104700A TW 102104700 A TW102104700 A TW 102104700A TW I506998 B TWI506998 B TW I506998B
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packet
network phone
proxy server
sip proxy
network
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TW102104700A
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TW201433128A (en
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Shaw Hwa Hwang
Bing Chih Yao
Chao Ping Chu
Ning Yun Ku
Tzu Hung Lin
Ming Che Yeh
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Univ Nat Taipei Technology
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Description

ICMP敏感性NAT之穿越法ICMP sensitive NAT traversal method

本發明有關於一種NAT(Network Address Translator)穿越法,尤指一種對於具有ICMP(Internet Control Message Protocol)敏感性的NAT之穿越法,其利用同步技術達成穿越。The present invention relates to a NAT (Network Address Translator) traversal method, and more particularly to a traversal method for NAT with ICMP (Internet Control Message Protocol) sensitivity, which uses synchronization technology to achieve traversal.

在目前的SIP(Session Initiation Protocol)網路環境中,架設NAT(Network Address Translator)伺服器十分普遍,但在NAT伺服器下的網路電話無法使語音封包直接對傳(Peer to Peer),必須透過SIP代理伺服器。In the current SIP (Session Initiation Protocol) network environment, it is very common to set up a Network Address Translator (NAT) server. However, a network phone under the NAT server cannot directly transmit a voice packet (Peer to Peer). Through the SIP proxy server.

網路電話為了使語音封包直接對傳,必須設法穿越NAT伺服器。相關的發明很多,例如中華民國發明專利I 376133號就是在電腦註冊階段發出一Invite訊息前,作多次的註冊程序,以偵知NAT伺服器分配通訊埠的規律變化,隨後的語音封包即以這種規律的變化預測所應分配的通訊埠而直接對傳,不必經過SIP代理伺服器。In order for the voicephone to directly transmit the voice packet, it must try to traverse the NAT server. There are many related inventions. For example, the Republic of China invention patent I 376133 is a registration procedure that is repeated several times before an Invite message is issued during the computer registration phase to detect the change of the protocol of the NAT server. The subsequent voice packet is This regular change predicts the communication that should be assigned and directly transmits it without having to go through the SIP proxy server.

但是上述這種穿越法碰到了具有ICMP(Internet Control Message Protocol)敏感性的NAT就失效了,說明如下:請見圖1,為SIP(Session Initiation Protocol)網路環境的語音封包直接傳輸示意圖,網路電話1與網路電話2分別在對稱型NAT3與對稱型 NAT4之下。語音封包中包含四個參數,即來源IP位址、來源通訊埠號碼、目的IP位址、目的通訊埠號碼。網路電話1發出語音封包-1給網路電話2,經過對稱型NAT3時,語音封包-1變成語音封包-1' ,語音封包-1中的來源IP位址VIP1被改成RIP1,來源通訊埠號碼SP1被改成SP1' ,目的IP位址RIP2與目的通訊埠號碼DP1維持不變。同樣地,網路電話2發出語音封包-2給網路電話1,經過對稱型NAT4時,語音封包-2變成語音封包-2' ,語音封包-2中的來源IP位址VIP2被改成RIP2,來源通訊埠號碼SP2被改成SP2' ,目的IP位址RIP1與目的通訊埠號碼DP2維持不變。However, the above-mentioned traversal method fails to encounter the ICMP (Internet Control Message Protocol)-sensitive NAT. The description is as follows: See Figure 1 for the direct transmission of voice packets in the SIP (Session Initiation Protocol) network environment. The road telephone 1 and the network telephone 2 are respectively under the symmetric NAT3 and the symmetric NAT4. The voice packet contains four parameters, namely the source IP address, the source communication port number, the destination IP address, and the destination communication number. Internet phone 1 sends voice packet-1 to network phone 2. When symmetric NAT3, voice packet-1 becomes voice packet-1 ' , and source IP address VIP1 in voice packet-1 is changed to RIP1, source communication The 埠 number SP1 is changed to SP1 ' , and the destination IP address RIP2 and the destination communication number DP1 remain unchanged. Similarly, the network phone 2 sends a voice packet-2 to the network phone 1. When the symmetric NAT4 passes, the voice packet-2 becomes the voice packet-2 ' , and the source IP address VIP2 in the voice packet-2 is changed to the RIP2. The source communication port number SP2 is changed to SP2 ' , and the destination IP address RIP1 and the destination communication port number DP2 remain unchanged.

雙方的語音封包若欲順利送達對方,必須:SP1' =DP2且DP1=SP2' 。SP1' 是由對稱型NAT3指定,SP2' 由對稱型NAT4指定,NAT伺服器指定或分配通訊埠是按規律變化的。利用中華民國發明專利I 376133號在電腦註冊階段發出一Invite訊息前,作多次的註冊程序,就可以偵知NAT伺服器分配通訊埠的規律變化,隨後的語音封包即以這種規律的變化預測所應分配的通訊埠而直接對傳。但即使直接對傳,雙方也不見得能完全同步對傳。If the voice packets of both parties are to be delivered to the other party, it must be: SP1 ' =DP2 and DP1=SP2 ' . SP1 ' is specified by symmetric NAT3, SP2 ' is specified by symmetric NAT4, and the NAT server specifies or allocates the communication 埠 to change according to the law. By using the Republic of China invention patent I 376133 to issue an Invite message during the computer registration phase, multiple registration procedures can be used to detect changes in the distribution of communication messages by the NAT server. The subsequent voice packets are changed by this law. The communication that should be assigned is predicted to be directly transmitted. But even if it is directly transmitted, the two sides will not be able to fully synchronize the pass.

若對稱型NAT3與對稱型NAT4是具有ICMP(Internet Control Message Protocol)敏感性的NAT(即對於駭客或病毒具有防禦性),當雙方語音封包不是完全同步對傳時,就會產生封鎖通訊埠的情況。請見圖2,網路電話1的語音封包-1經對稱型NAT3轉成語音封包-1' 送入對稱型NAT4,此時若雙方語音封包不是完全同步對傳,對稱型NAT4的通訊埠尚未打開,對稱型NAT4就會產生防禦性動作,回送一ICMP拒絕性封包給對稱型NAT3,對稱型NAT3因此自我封鎖通訊埠(如◆號所示),當網路電話2送出一語音封包 -2經對稱型NAT4轉成語音封包-2' 送入對稱型NAT3時,就被對稱型NAT3擋住。此稱為外鎖(outbound)。If symmetric NAT3 and symmetric NAT4 are ICMP (Internet Control Message Protocol) sensitive NAT (that is, defensive for hackers or viruses), when both voice packets are not completely synchronized, a blocked communication will occur. Case. Please refer to FIG. 2, the voice packet-1 of the network telephone 1 is converted into a voice packet -1 ' transformed into a symmetric NAT4 by symmetric NAT3. At this time, if the voice packets of both parties are not completely synchronized, the communication of the symmetric NAT4 has not yet been communicated. Open, symmetric NAT4 will generate defensive actions, return an ICMP reject packet to symmetric NAT3, symmetric NAT3 thus self-block communication (as indicated by the number ◆), when the network phone 2 sends a voice packet-2 When symmetric NAT4 is converted into voice packet-2 ' when it is sent to symmetric NAT3, it is blocked by symmetric NAT3. This is called outbound.

另一種情況請見圖3,網路電話1的語音封包-1經對稱型NAT3轉成語音封包-1' 送入對稱型NAT4,此時若雙方語音封包不是完全同步對傳,對稱型NAT4的通訊埠尚未打開,對稱型NAT4就會產生防禦性動作,回送一ICMP拒絕性封包給對稱型NAT3,對稱型NAT4同時自我封鎖通訊埠(如◆號所示),當網路電話B送出一語音封包-2經對稱型NAT4時,就被對稱型NAT4擋住。此稱為內鎖(inbound)。Another case where voice packets see Figure 3, an Internet telephone via the symmetrical NAT3 -1 converted into voice packets 1 'into symmetrical NAT 4, at this time if the two sides of the voice packets are not fully synchronized transmission, the symmetric NAT 4 The communication port is not yet open, the symmetric NAT4 will generate defensive actions, and send an ICMP reject packet to the symmetric NAT3. The symmetric NAT4 also self-blocks the communication port (as shown by the ◆), when the network phone B sends a voice. When packet-2 passes through symmetric NAT4, it is blocked by symmetric NAT4. This is called inbound.

為了解決ICMP敏感性的NAT遇到雙方語音封包不是完全同步對傳時,就會產生封鎖通訊埠的情況,本發明在SIP通訊協定中加上一同步階段以解決穿越的問題。In order to solve the problem that the ICMP-sensitive NAT encounters that the voice packets of the two parties are not completely synchronized, the communication is blocked. The present invention adds a synchronization phase to the SIP protocol to solve the problem of traversal.

本發明將SIP通訊協定分成註冊階段、通訊埠預測階段、同步階段與媒體階段,其網路環境由一第一網路電話、一第二網路電話、一第一對稱型NAT、一第二對稱型NAT與一SIP代理伺服器組成,一第一網路電話在第一對稱型NAT之下,一第二網路電話在第二對稱型NAT之下,第一對稱型NAT與第二對稱型NAT均具有ICMP敏感性,該穿越法包含:第一網路電話與第二網路電話先向SIP代理伺服器註冊,完成註冊階段;第一網路電話向第一對稱型NAT作多次(N次)偵測程序,以偵知第一對稱型NAT分配通訊埠的規律變化;第二網路電話向第二對稱型NAT作多次(N次)偵測程序,以偵知第二對稱型NAT分配通訊埠的規律變 化,完成通訊埠預測階段;此後進入同步階段,第一網路電話發出一試驗性封包經T1時間到達SIP代理伺服器,SIP代理伺服器將該封包回傳至第一網路電話,歷經T2時間;(T1+T2)除以2即為第一網路電話送封包到達SIP代理伺服器的平均時間T3,此後第一網路電話發送一封含有時間資訊T3的封包到達SIP代理伺服器,由SIP代理伺服器保留住;同樣地,第二網路電話發出一試驗性封包經T4時間到達SIP代理伺服器,SIP代理伺服器將該封包回傳至第二網路電話,歷經T5時間;(T4+T5)除以2即為第二網路電話送封包到達SIP代理伺服器的平均時間T6,此時第二網路電話發送一封含有時間資訊T6的封包到達SIP代理伺服器,由SIP代理伺服器保留住;SIP代理伺服器收下了雙方含有時間資訊的封包後,接著就將含有時間資訊的封包同時分別交換傳給對方,第一網路電話將會收到由第二網路電話發送含有時間資訊T6的封包,並取得T6;而第二網路電話將會收到由第一網路電話發送含有時間資訊T3的封包,並取得T3;若T6>T3,則第二網路電話在收到封包後即可直接進入媒體階段,不經SIP代理伺服器,直接發送語音封包給第一網路電話;第一網路電話在收到封包後尚須等待T6-T3時間,才可直接進入媒體階段,不經SIP代理伺服器,直接發送語音封包給第二網路電話,此後雙方進入媒體階段,直接同步互傳語音封包;若T3>T6,則第一網路電話在收到封包後即可直接進入媒體階段,不經SIP代理伺服器,直接發送語音封包給第二網路電話;第二網路 電話在收到封包後尚須等待T3-T6時間,才可直接進入媒體階段,不經SIP代理伺服器,直接發送語音封包給第一網路電話,此後雙方進入媒體階段,直接同步互傳語音封包。The invention divides the SIP communication protocol into a registration phase, a communication, a prediction phase, a synchronization phase and a media phase, and the network environment is composed of a first network phone, a second network phone, a first symmetric NAT, and a second The symmetric NAT is composed of a SIP proxy server, a first network phone is under the first symmetric NAT, a second network phone is under the second symmetric NAT, the first symmetric NAT and the second symmetric Type NAT has ICMP sensitivity. The traversal method includes: the first network phone and the second network phone first register with the SIP proxy server to complete the registration phase; the first network phone makes multiple times to the first symmetric NAT. (N times) detection procedure to detect the regular change of the first symmetric NAT distribution communication protocol; the second network telephone makes multiple (N times) detection procedures to the second symmetric NAT to detect the second The regularity of symmetric NAT distribution communication After completing the communication/predicting phase; after entering the synchronization phase, the first network phone sends a test packet to the SIP proxy server via the T1 time, and the SIP proxy server returns the packet to the first network phone, after T2. Time; (T1+T2) divided by 2 is the average time T3 that the first network phone sends the packet to the SIP proxy server, after which the first network phone sends a packet containing the time information T3 to the SIP proxy server. The SIP proxy server retains; in the same manner, the second network phone sends a test packet to the SIP proxy server via the T4 time, and the SIP proxy server returns the packet to the second network phone for T5 time; (T4+T5) divided by 2 is the average time T6 when the second network phone sends the packet to the SIP proxy server. At this time, the second network phone sends a packet containing the time information T6 to the SIP proxy server. The SIP proxy server retains; after the SIP proxy server receives the packets containing the time information, the packets containing the time information are simultaneously exchanged and transmitted to the other party, and the first network phone will receive the second packet. The road phone sends the packet containing the time information T6 and obtains T6; and the second network phone receives the packet containing the time information T3 sent by the first network phone, and obtains T3; if T6>T3, then the second After receiving the packet, the network phone can directly enter the media stage, and directly send the voice packet to the first network phone without the SIP proxy server; the first network phone must wait for the T6-T3 time after receiving the packet. Only then can enter the media stage directly, without sending a voice packet to the second network phone without the SIP proxy server, after which the two parties enter the media phase and directly synchronize the voice packets; if T3>T6, the first network phone After receiving the packet, it can directly enter the media stage, and directly send the voice packet to the second network phone without the SIP proxy server; the second network After receiving the packet, the phone still has to wait for T3-T6 time before entering the media stage directly, and directly sends the voice packet to the first network phone without the SIP proxy server. After that, the two parties enter the media phase and directly synchronize the voice transmission. Packet.

1‧‧‧網路電話1‧‧‧Internet phone

2‧‧‧網路電話2‧‧‧Internet phone

3‧‧‧對稱型NAT3‧‧‧Symmetric NAT

4‧‧‧對稱型NAT4‧‧‧Symmetric NAT

5‧‧‧SIP代理伺服器5‧‧‧SIP proxy server

圖1為SIP網路環境的語音封包傳輸示意圖。FIG. 1 is a schematic diagram of voice packet transmission in a SIP network environment.

圖2為外鎖(outbound)示意圖。Figure 2 is a schematic diagram of the outbound.

圖3為內鎖(inbound)示意圖。Figure 3 is a schematic diagram of an inbound.

圖4為註冊階段、通訊埠預測階段示意圖。Figure 4 is a schematic diagram of the registration phase and communication prediction phase.

圖5為同步階段、媒體階段示意圖。Figure 5 is a schematic diagram of the synchronization phase and the media phase.

本發明將SIP網路環境分成四個階段,即註冊階段(Login Session)、通訊埠預測階段(Port Prediction Session)、同步階段(Synchronization Session)與媒體階段(Media Session)。The present invention divides the SIP network environment into four phases, namely, a Login Session, a Port Prediction Session, a Synchronization Session, and a Media Session.

註冊階段、通訊埠預測階段請見圖4。圖5接續圖4,示出同步階段與媒體階段。Please refer to Figure 4 for the registration phase and communication forecasting phase. Figure 5 continues with Figure 4, showing the synchronization phase and the media phase.

圖4中,網路電話1與網路電話2先向SIP代理伺服器5註冊,完成註冊階段。In FIG. 4, the network telephone 1 and the network telephone 2 first register with the SIP proxy server 5 to complete the registration phase.

接著進入通訊埠預測階段,網路電話1以Register指令作多次(N次)偵測程序,以偵知對稱型NAT3分配通訊埠的規律變化。在做完N次偵測程序後,網路電話1在傳輸語音封包時,便會依據對稱型NAT3分配通訊埠的規律變化,預測出對稱型NAT3所將分配的通訊埠號碼而利用作為傳輸語音封包的管道。Then enter the communication 埠 prediction phase, the VoIP phone 1 uses the Register command to make multiple (N times) detection procedures to detect the regular change of the symmetric NAT3 distribution communication 埠. After completing the N detection procedure, when the voice packet is transmitted by the network telephone 1, the regular change of the communication port is allocated according to the symmetric NAT3, and the communication port number to be assigned by the symmetric NAT3 is predicted to be used as the transmission voice. The pipe of the packet.

網路電話1接著發出New Invite訊息經過對稱型NAT3傳送給SIP代理伺服器5,SIP代理伺服器5則發出New Invite-1訊息經過對稱型NAT4轉送給網路電話2。The VoIP 1 then sends a New Invite message to the SIP Proxy Server 5 via symmetric NAT3, and the SIP Proxy Server 5 sends a New Invite-1 message to the VoIP 2 via the symmetric NAT4.

網路電話2接收到New Invite-1訊息後,以Register指令作多次(N次)偵測程序,以偵知對稱型NAT4分配通訊埠的規律變化。在作完N次偵測程序之後,網路電話2在傳輸語音封包時,便會依據對稱型NAT4分配通訊埠的規律變化,預測出對稱型NAT4所將分配的通訊埠號碼而利用作為傳輸語音封包的管道。至此完成通訊埠預測階段。通訊埠預測階段的方法不限於此法,此法只是舉例說明。After receiving the New Invite-1 message, the VoIP phone 2 uses the Register command to perform multiple (N times) detection procedures to detect the regular change of the symmetrical NAT4 distribution communication port. After the N detection process is completed, the network phone 2 transmits the voice packet according to the regular change of the communication port of the symmetric NAT4, and predicts the communication port number to be allocated by the symmetric NAT4 as the transmission voice. The pipe of the packet. This completes the communication/predictive phase. The method of communication prediction stage is not limited to this method, and this method is only an example.

請見圖5,說明同步階段與媒體階段。網路電話1先發一試驗性封包經T1時間到達SIP代理伺服器5,SIP代理伺服器5將該封包回傳至網路電話1,歷經T2時間。(T1+T2)除以2即為網路電話1送封包到達SIP代理伺服器5的平均時間T3,此時網路電話1發送一封含有時間資訊T3的封包到達SIP代理伺服器5,由SIP代理伺服器5保留住。See Figure 5 for the synchronization phase and media phase. The network phone 1 first sends a test packet to the SIP proxy server 5 via the T1 time, and the SIP proxy server 5 transmits the packet back to the network phone 1, after T2. (T1+T2) divided by 2 is the average time T3 when the VoIP phone 1 sends the packet to the SIP proxy server 5. At this time, the VoIP phone 1 sends a packet containing the time information T3 to the SIP proxy server 5, by The SIP proxy server 5 is retained.

同樣地,網路電話2先發一試驗性封包經T4時間到達SIP代理伺服器5,SIP代理伺服器5將該封包回傳至網路電話2,歷經T5時間。(T4+T5)除以2即為網路電話2送封包到達SIP代理伺服器5的平均時間T6,此時網路電話2發送一封含有時間資訊T6的封包到達SIP代理伺服器5,由SIP代理伺服器5保留住。Similarly, the network phone 2 first sends a test packet to the SIP proxy server 5 via the T4 time, and the SIP proxy server 5 transmits the packet back to the network phone 2 for T5. (T4+T5) divided by 2 is the average time T6 when the VoIP phone 2 sends the packet to the SIP proxy server 5, at this time, the VoIP phone 2 sends a packet containing the time information T6 to the SIP proxy server 5, by The SIP proxy server 5 is retained.

SIP代理伺服器收下了雙方含有時間資訊的封包後,接著就將含有時間資訊的封包同時分別交換傳給對方,網路電話1將會收到由網路電話2發送含有時間資訊T6的封包,並取得T6;而網路電話2將會收到由網 路電話1發送含有時間資訊T3的封包,並取得T3。After receiving the packet with time information on both sides, the SIP proxy server then exchanges the packets containing the time information to the other party at the same time, and the network phone 1 will receive the packet containing the time information T6 sent by the network phone 2. And get T6; while VoIP 2 will receive the net The road telephone 1 transmits a packet containing time information T3 and acquires T3.

若T6>T3,則網路電話2在收到封包後即可直接進入媒體階段,不經SIP代理伺服器5,直接發送語音封包給網路電話1。網路電話1在收到封包後尚須等待T6-T3時間,才可直接進入媒體階段,不經SIP代理伺服器5,直接發送語音封包給網路電話2。此後雙方進入媒體階段,直接同步互傳語音封包。If T6>T3, the network phone 2 can directly enter the media stage after receiving the packet, and directly send the voice packet to the network phone 1 without the SIP proxy server 5. After receiving the packet, the VoIP phone 1 has to wait for the T6-T3 time to enter the media phase directly, and directly sends the voice packet to the VoIP phone 2 without the SIP proxy server 5. After that, the two parties entered the media stage and directly synchronized the voice packets.

若T3>T6,則網路電話1在收到封包後即可直接進入媒體階段,不經SIP代理伺服器5,直接發送語音封包給網路電話2。網路電話2在收到封包後尚須等待T3-T6時間,才可直接進入媒體階段,不經SIP代理伺服器5,直接發送語音封包給網路電話1。此後雙方進入媒體階段,直接同步互傳語音封包。If T3>T6, the network phone 1 can directly enter the media stage after receiving the packet, and directly send the voice packet to the network phone 2 without the SIP proxy server 5. After receiving the packet, the VoIP phone 2 has to wait for the T3-T6 time to enter the media phase directly, and directly sends the voice packet to the VoIP phone 1 without the SIP proxy server 5. After that, the two parties entered the media stage and directly synchronized the voice packets.

本發明的精神與範圍決定於下面的申請專利範圍,不受限於上述實施例。The spirit and scope of the present invention are determined by the scope of the following claims, and are not limited to the above embodiments.

1‧‧‧網路電話1‧‧‧Internet phone

2‧‧‧網路電話2‧‧‧Internet phone

3‧‧‧對稱型NAT3‧‧‧Symmetric NAT

4‧‧‧對稱型NAT4‧‧‧Symmetric NAT

5‧‧‧SIP代理伺服器5‧‧‧SIP proxy server

Claims (1)

一種ICMP敏感性NAT之穿越法,將一SIP通訊協定分成註冊階段、通訊埠預測階段、同步階段與媒體階段,其網路環境由一第一網路電話、一第二網路電話、一第一對稱型NAT、一第二對稱型NAT與一SIP代理伺服器組成,一第一網路電話在第一對稱型NAT之下,一第二網路電話在第二對稱型NAT之下,第一對稱型NAT與第二對稱型NAT均具有ICMP敏感性,該穿越法包含:a.第一網路電話與第二網路電話先向SIP代理伺服器註冊,完成註冊階段;b.第一網路電話向第一對稱型NAT作多次(N次)偵測程序,以偵知第一對稱型NAT分配通訊埠的規律變化;第二網路電話向第二對稱型NAT作多次(N次)偵測程序,以偵知第二對稱型NAT分配通訊埠的規律變化,完成通訊埠預測階段;c.此後進入同步階段,第一網路電話發出一試驗性封包經T1時間到達SIP代理伺服器,SIP代理伺服器將該封包回傳至第一網路電話,歷經T2時間;(T1+T2)除以2即為第一網路電話送封包到達SIP代理伺服器的平均時間T3,此後第一網路電話發送一封含有時間資訊T3的封包到達SIP代理伺服器,由SIP代理伺服器保留住;d.同樣地,第二網路電話發出一試驗性封包經T4時間到達SIP代理伺服器,SIP代理伺服器將該封包回傳至第二網路電話,歷經T5時間;(T4+T5)除以2即為第二網路電話送封包到達SIP代理 伺服器的平均時間T6,此時第二網路電話發送一封含有時間資訊T6的封包到達SIP代理伺服器,由SIP代理伺服器保留住;e. SIP代理伺服器收下了雙方含有時間資訊的封包後,接著就將含有時間資訊的封包同時分別交換傳給對方,第一網路電話將會收到由第二網路電話發送含有時間資訊T6的封包,並取得T6;而第二網路電話將會收到由第一網路電話發送含有時間資訊T3的封包,並取得T3;f.若T6>T3,則第二網路電話在收到封包後即可直接進入媒體階段,不經SIP代理伺服器,直接發送語音封包給第一網路電話;第一網路電話在收到封包後尚須等待T6-T3時間,才可直接進入媒體階段,不經SIP代理伺服器,直接發送語音封包給第二網路電話,此後雙方進入媒體階段,直接同步互傳語音封包;g.若T3>T6,則第一網路電話在收到封包後即可直接進入媒體階段,不經SIP代理伺服器,直接發送語音封包給第二網路電話;第二網路電話在收到封包後尚須等待T3-T6時間,才可直接進入媒體階段,不經SIP代理伺服器,直接發送語音封包給第一網路電話,此後雙方進入媒體階段,直接同步互傳語音封包。An ICMP-sensitive NAT traversal method divides a SIP communication protocol into a registration phase, a communication, a prediction phase, a synchronization phase, and a media phase, and the network environment is composed of a first network phone, a second network phone, and a first network phone. A symmetric NAT, a second symmetric NAT and a SIP proxy server, a first network phone is under the first symmetric NAT, and a second network phone is under the second symmetric NAT. A symmetric NAT and a second symmetric NAT both have ICMP sensitivity, and the traversal method includes: a. The first network phone and the second network phone first register with the SIP proxy server to complete the registration phase; b. The network phone makes multiple (N times) detection procedures to the first symmetric NAT to detect the regular change of the first symmetric NAT distribution communication; the second network telephone makes multiple times to the second symmetric NAT ( N times) detection procedure to detect the change of the second symmetric NAT distribution communication protocol, complete the communication埠 prediction phase; c. After entering the synchronization phase, the first network phone sends a test packet to reach the SIP through the T1 time. Proxy server, SIP proxy server returns the packet to the first An Internet phone, after T2 time; (T1+T2) divided by 2 is the average time T3 of the first network phone to send the packet to the SIP proxy server, after which the first network phone sends a message containing time information T3. The packet arrives at the SIP proxy server and is retained by the SIP proxy server; d. Similarly, the second network phone sends a test packet to the SIP proxy server via the T4 time, and the SIP proxy server returns the packet to the first Two network calls, after T5 time; (T4 + T5) divided by 2 is the second network phone to send packets to the SIP proxy The average time of the server is T6. At this time, the second network phone sends a packet containing the time information T6 to the SIP proxy server, which is retained by the SIP proxy server; e. The SIP proxy server receives the time information of both parties. After the packet, the packet containing the time information is simultaneously exchanged and transmitted to the other party, and the first network phone will receive the packet containing the time information T6 sent by the second network phone, and obtain the T6; The road phone will receive the packet containing the time information T3 sent by the first network phone, and obtain T3; f. If T6>T3, the second network phone can directly enter the media stage after receiving the packet, The SIP proxy server directly sends the voice packet to the first network phone; after receiving the packet, the first network phone has to wait for the T6-T3 time before entering the media phase directly, without the SIP proxy server, directly Sending a voice packet to the second network phone, after which the two parties enter the media phase and directly synchronize the voice packets to each other; g. If T3>T6, the first network phone can directly enter the media stage after receiving the packet, without SIP proxy The server directly sends the voice packet to the second network phone; after receiving the packet, the second network phone has to wait for the T3-T6 time before entering the media phase directly, and directly sending the voice packet without the SIP proxy server. The first network phone is sent, and then both parties enter the media stage to directly synchronize the voice packets.
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