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Figure 5-32 A call from H323 to SIP using H323 Fast Connect
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Gateway
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Setup faststart [logical chan info = G711 TX, G711 RX 1236789:2000
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INVITE To: User@SIPnet c=IN IP4 1236789 m=audio 2000 RTP/AVP 0
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c 180 (Ringing) d Alerting e f Connect faststart [logical chan info = G711 TX, G711 RX 1234567:8000 g ACK h Two-way voice Two-way voice 200 (OK) To: User@SIPnet c=IN IP4 1234567 m=audio 8000 RTP/AVP 0
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The Session Initiation Protocol (SIP)
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5
Gateway SIP Client
Figure 5-33 A call from H323 to SIP without H323 Fast Connect
a Setup (no faststart) b
INVITE (default message body a=inactive) c 200 (OK) d Connect (no faststart) e Terminal Capability Set G711 TX, G711 RX f Terminal Capability Set Ack g Terminal Capability Set G711 TX, G711 RX h Terminal Capability Set Ack i Open Logical Channel j k Open Logical Channel Ack RX=1236789:2000 UDATE c=IN IP4 1236789 m=audio 2000 RTP/AVP 0 a= sendrecv 200 (OK) m Open Logical Channel Ack RX=1234567:8000 n Two-way voice Two-way voice c=IN IP4 1234567 m=audio 8000 RTP/AVP 0 H245 SIP H2250 ACK c=IN IP4 1234567 m=audio 8000 RTP/AVP 0
l Open Logical Channel
In that event, the INVITE could contain a default session description The IP address in the connection (c ) line of the session description could contain a default value, while the media (m) line would indicate a port number of 0 and an attribute of inactive, as shown in Figure 5-33 Once the appropriate H245 signaling has taken place between the gateway and the
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The Session Initiation Protocol (SIP)
The Session Initiation Protocol (SIP)
H323 terminal, the gateway can issue a SIP UPDATE to the SIP client, including a revised session description
Summary
Although quite new, SIP has already established itself as the way of the future for signaling in VoIP networks Its strengths include the fact that it is far simpler than H323 and is therefore easier to implement Its simplicity, however, belies its flexibility and power It can support all the features that are already supported by traditional telephony networks and more besides In fact, SIP is the protocol of choice for the evolution of thirdgeneration wireless networks, where it is expected that SIP-based mobile devices will become available and SIP-based network elements will be introduced within mobile networks As we shall see in the next chapter, SIP also fits very well with those protocols used for media gateway control and, as such, forms part of the overall architecture known as softswitch
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The Session Initiation Protocol (SIP)
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Source: Carrier Grade Voice Over IP
CHAPTER
Media Gateway Control and the Softswitch Architecture
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Media Gateway Control and the Softswitch Architecture
6
Previous chapters described some of the advantages that Voice over IP (VoIP) can offer over traditional circuit-switched telephony, such as lower cost of network implementation, integration of voice and data applications, new features, and potentially reduced bandwidth for voice calls It would be nice if all telephony were to be carried over the Internet Protocol (IP) so that these advantages could be made available globally Unfortunately, that day will not arrive for a long time Replacing all traditional circuit-switched networks is not feasible, and perhaps not even desirable Among other reasons, the cost would simply be exorbitant Rather, one can expect a gradual evolution from circuit-switched to IP-based networks such that both types of networks will exist side by side for a very long time Given that VoIP networks and traditional circuit-switched networks will have to coexist for many years, they will need to interwork as seamlessly as possible In other words, users of existing circuit-switched systems should be able to place calls to VoIP users, and vice versa In each case, the user should not have to do anything radically different from simply calling another user on the same type of network The challenge therefore is to develop solutions that make such seamless interworking possible
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