In basic SIP session setup, a SIP UA client (UAC) will sends an invite request to the SIP URL of the endpoint which is UAS. If IP address of the UAS is known to UAC, it will directly send the request. But if the IP address of the UAS is not known then UAC will sends the request to proxy server or redirect server to locate the UAS. As SIP address is mapped to an IP address, call request will be forwarded to the UAS. As UAS accepts the call request, the call will be successful otherwise it can be diverted to voice mail.
Call flow between two SIP Gateways:
SIP gateways are useful when call has been requested from non-SIP phone. In such cases SIP gateways will function as SIP UAs and initiates the SIP session between two end users. Figure X shows how two routers handle the analog phones using SIP between them. Here PBX will send a request for call setup to GATEWAY-1 using normal analog call signaling. SIP GATEWAY-1 will acts as a UAC and sends INVITE request to GATEWAY-2 which is acting as UAS. Here only two gateways will exchange the SIP messages.
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As the analog phone initiates the call, the call flow will be as below:
The PBX (Private Branch Exchange) sends a call setup to GATEWAY-1. As it receives a call setup request, a SIP INVITE message will be sent to GATEWAY-2 from GATEWAY-1 via IP networks and it also sends a Call Processing message to PBX. SIP INVITE message contains SDP information for capabilities negotiation.
Call setup messages will be exchanged between GATEWAY-2 and PBX. And SIP Response 100 message will be sent to GATEWAY-1. As GATEWAY-2 receives alerting message from its PBX, it will send SIP 180 message to GATEWAY-1.
GATEWAY-1 will notify it's PBX about messages-came from GATEWAY-2 using - using analog signaling.
As the receiver picks up the call, his PBX will sends a Connect message to GATEWAY-2 and then GATEWAY-2 will sends a 200 OK message response to GATEWAY-1 and it will contains SDP information which will mention the capabilities supported by both devices.
After receiving SIP 200, GATEWAY-1 will send a Connect message to its PBX. When GATEWAY-1 will receive an acknowledgement from its PBX, a SIP ACK message is sent to GATEWAY-2.
A Connect acknowledgement is sent to GATEWAY-2's PBX. After this call is active. At this active call stage a RTP streams are exists between gateways normal voice streams are exists between the two analog phones.
When caller hangs up the call, a Call Disconnect message will be sent to GATEWAY-1 and then GATEWAY-1 will sends a SIP BYE message to GATEWAY-2. Release and Release Complete messages will be exchanged between GATEWAY-1 and its PBX.
Similarly Call Disconnect and Release messages will be exchanged between GATEWAY-2 and its PBX.
To terminate the call completely, GATEWAY-2 will send SIP 200 OK message to GATEWAY-1 and also sends a Release message to its own PBX.
Call Flow Using Proxy Server.
SIP Proxy servers are used when end points are like: a Computer running a SIP application, a SIP phone, and a cell phone that uses SIP. Proxy servers act as intermediary for SIP calls.
Every SIP UA has to register itself with a proxy server. A special Record route option is available in proxy server. When this option is enable it will stay in communication link between UAC and gateway and knows the status of a call otherwise it will just leave the communication link and rest of the communication will occurs directly between UAC and gateway. Figure Y shows the call flow when SIP Proxy Servers are used.
In figure Y, one endpoint requests a call to an analog phone. The call flow proceeds as bellow:
An INVITE message with SDP information will be sent to proxy server for initiating the call. Phone number of a destination end will be saved in the Request URI field of the message.
Proxy server will creates a new INVITE message by copying the information from the previous message but it will put address of GATEWAY-2 in the Request URI field.
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GATEWAY-2 exchanges call setup message with its PBX and also sends SIP Response 100 (trying) to proxy server. And then proxy server will send response 100 (Trying) to SIP UAC which is not mandatory.
PBX sets up the call with its analog call and sends alerting message to GATEWAY-2 and then GATEWAY-2 will sends a SIP 180 Ringing message to proxy server. Proxy server will forward this message to UA Client.
As receiver picks up the phone, a Connect message is being sent to GATEWAY-2 from PBX and then GATEWAY-2 will sends response message 200 OK to proxy server. Proxy server will forward this message to UA Client. This SIP message will contain SDP information for the session. After this proxy server will leaves the signaling path as the Record-route is disabled. Further communication will be between UA Client and GATEWAY-2.
UAC sends a n acknowledgement to GATEWAY-2 and GATEWAY-2 and then GATEWAY-2 sends Connect ACK to PBX. As the call become active a RTP stream exists between the GATEWAY-2 and UAC. And normal voice stream exists between GATEWAY-2 and PBX.
When endpoint hangs up it will send BYE message to GATEWAY-2 and GATEWAY-2 will send disconnect message to PBX.
PBX response with release message to GATEWAY-2. GATEWAY-2 will send 200 OK message to UA Client and Release Complete message to PBX. Call is terminated completely at this point.
Call Flow Using Multiple Servers
Multiple servers are required for making a call to the number which is outside the local domain. For making a call outside the local domain proxy server, redirect server and registrar server will act for completing the call. Here proxy server requests the details to redirect server about where to send an INVITE message. In reply, redirect server will give information about endpoint address or address of next hop server. On basis of this information, proxy server will route the INVITE message.
The call flow using proxy server, redirect server and registrar server is shown in figure z. These servers can be in one device as they are just functional components . Figure shows how INVITE message is routed using different servers.
Change proxy A - > proxy C and proxy b ïƒ proxy s, network a,b ïƒ 1,2
As the SIP device initiates the INVITE message, Call flow will take place as follow:
GATEWAY-2 will first send register message to registrar sever for registering its analog phone. Registrar server replies with 200 OK message which indicate that phone number has been registered now.
As any SIP device try to initiates the call it will send INVITE message to its proxy server -C.
On receiving the INVITE message, proxy server finds that requested number is not in local domain. So it will forward this INVITE message to redirect server for getting information for routing the message.
In response, Redirect server will send 300- series message which contain the details about the SIP address of Proxy server S.
Proxy server C has now information that next hop for routing this INVITE message is Proxy Server S. So it will forward the INVITE message to Proxy Server S.
Proxy Server S now sends a query message to registrar server for the location of the destination number. In response message Registrar server provides the SIP address of GATEWAY-2.
Now Proxy Server S knows the address of called number. It will send INVITE message to GATEWAY-2.
Now GATEWAY-2 will set up a call with its PBX. After setting up the call it will notify to Proxy server S. This notification will be forwarded to endpoint via the Proxy servers. Remaining call flow will take place as described in section X.X.