- When a UA desires to establish a multimedia session, it sends an INVITE method to the remote UA. In the following diagram, you can see an example of a basic call flow:
An end device in SIP is called UA. A user agent can initiate or receive a call. A UA can be an IP phone, video camera, software client, or any device or software that participates in an SIP session.
- After INVITE, you should see Trying, Session Progress, Ringing, or a combination of them coming from the other side
- We can see here how the session progress between the initiator on 172.18.110.200 and the responder on 172.18.110.203:
- The INVITE method is sent from the session initiator. This will always be the first packet that starts the conversation.
- The responder answers with Trying (code 100), Session progress (code 183), and after three seconds with Ringing (code 180). Then it answers with OK (code 200), meaning that the handset was picked up.
- In the preceding topology, there are multiple CUBEs in between that help to establish end to end call flow
- The CUBE or switch 172.18.110.200 on the left sends an INVITE request to switch 172.18.110.203
- The switch 172.18.110.203 replies with the SIP Trying message.
- The switch sends an INVITE to the CUBE or switch 172.18.110.206 on the right.
- The switch 172.18.110.206 sends Trying (code 100), and then the session progresses (code 183) to the switch.
- When 172.18.110.203 receives Ringing from 172.18.110.206, it in turn sends Ringing message to 172.18.110.200.
- The destination endpoint sends SIP 200 OK to the communications manager at 172.18.110.206 when the call is answered. The SIP 200 OK message carries SDP content in the message body. The SDP provides information about RTP UDP port number and the list audio and video codec offered by the destination end point (also referred to as SDP Offer). The SIP 200 OK message traverses SBC (172.18.110.203), communication manager (172.18.110.200) and reaches the origination endpoint.
- In the preceding sample screenshot, the SDP message instructs to use UDP port 25944 for the RTP audio stream. In addition, it includes other details such as the codec supported.
- The origination endpoint selects one of the audio codes and sends the selected codec and its RTP port number information (also referred to as SDP answer) in the SIP ACK to its communication manager (172.18.110.200). This SIP ACK message traverses SBC (172.18.110.203), communication manager (172.18.110.206) and reaches the destination endpoint. Upon receiving the ACK, the destination endpoint starts sending RTP packets at port 8260. It will also receive RTP packets at port 25944.
- When the user hangs up the call, SIP BYE message will be exchanged between the SIP devices to terminate the call signaling session.
- If an error message is received at any stage, the connection will not be established.
Don't forget that SIP works over UDP. And since UDP does not open any connection to the other side before sending the request, it can be possible that a request will not arrive to the destination simply because of a network reachability problem. For this reason, when you don't get a response, it could be that the INVITE simply didn't get to the destination because of a network problem.
SIP error codes are listed in the following tables, along with their possible reasons. Unless mentioned otherwise, the codes are defined in RFC 3261.