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SIP VOIP Session Call Flow

SIP VOIP Session Call Flow outlines the key steps in establishing and terminating a SIP VOIP call: 1. Registration - A softphone registers with a SIP server to be found for calls. 2. Call Establishment - The caller sends an Invite, the callee responds with rings and an ok, and call parameters are negotiated. 3. VOIP Call - Audio and video data is transmitted between users via RTP during the call. 4. Call Termination - Either user can end the call by sending a Bye request, confirmed by a 200 response.

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176 views

SIP VOIP Session Call Flow

SIP VOIP Session Call Flow outlines the key steps in establishing and terminating a SIP VOIP call: 1. Registration - A softphone registers with a SIP server to be found for calls. 2. Call Establishment - The caller sends an Invite, the callee responds with rings and an ok, and call parameters are negotiated. 3. VOIP Call - Audio and video data is transmitted between users via RTP during the call. 4. Call Termination - Either user can end the call by sending a Bye request, confirmed by a 200 response.

Uploaded by

Baraja Online
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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SIP VOIP Session Call Flow

A: Registration

When a user agent (say a softphone) launches, it needs to register with a SIP server in order to be
found by other user agents. The SIP Register request message is used for this. It provides the
location bindings through the To and From SIP URIs. Optionally, an additional binding can be
provided through the Contact field.

SIP Register Message


REGISTER sip:sip.cybercity.dk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2;branch=z9hG4bKnp151248737-
46ea715e192.168.1.2;rport
From: <sip:voi18063@sip.cybercity.dk>;tag=903df0a
To: <sip:voi18063@sip.cybercity.dk>
Call-ID: 578222729-4665d775@578222732-4665d772
Contact:
<sip:voi18063@192.168.1.2:5060;line=9c7d2dbd8822013c>;expires=1200;q=0.500
Expires: 1200
CSeq: 68 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: Nero SIPPS IP Phone Version 2.0.51.16
B: Call Establishment

Call establishment is where the magic happens. There are a few steps here, so let's cover them
one-by-one in sequence.

1. SIP Invite Request - The SIP Invite starts the call establishment attempt. This message
contains the callee (SIP URI in the To field). This is sent from the caller to the SIP server
where it looks up the callee. In a larger network, the SIP server may need to consult other
SIP servers if the callee is not local. Once the callee is located, the Invite is forwarded.
For VOIP, the Invite also includes a SDP message body with the parameters for the VOIP
call.
2. SIP Response 100 (Trying) - This message is sent from the SIP server to the callee to
confirm the Invite request.
3. SIP Response 180 (Ringing) - This message indicates that the Invite was received by the
callee and their user agent is alerting the user.
4. SIP Response 200 (OK) - When the user picks up, a 200 response is sent back to
confirm the call. Additionally, the callee sends a SDP message body with its VOIP call
parameters. As a result of this message and the initial Invite from the caller, an exchange
and negotiation of VOIP call parameters has occurred.
5. SIP Ack Request - Finally, the caller confirms with an Ack request back to the callee.
The callee then initiates the VOIP call to the caller.

C: VOIP Call

The VOIP call itself is transmitted between the user agents using RTP (real-time transport
protocol). This protocol is used for delivering audio and video data over IP networks. An
additional protocol RTCP (RTP control protocol) is used to provide statistics and control for the
RTP transmissions. We will cover RTP and RTCP in an upcoming blog.

D: Call Termination

When a user decides to terminate a call, a SIP Bye request is sent. Either side of the call can
terminate. The other user agent then responds with a SIP 200 status code response to confirm the
termination.

https://forums.extrahop.com/t/sip-protocol-overview-performance-monitoring-extrahop/947

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