SIP Call Illustration

The protocols used in an IP phone call are:
SIP (Session Initiation Protocol):

A standardized signaling protocol (RFC 3261) which works over TCP
(typically on port 5060) at the application OSI layer. Its role is to
create, modify or terminate phone sessions.
SIP behaves very similarily to HTTP in that SIP clients send requests
to the server which will answer with responses (status). The
difference with HTTP is that SIP clients can also respond to requests
made by a server.
Other signaling protocols are H.323 or the Cisco protocol SCCP. SIP
is progressively replacing these two protocols.
SDP (Session Description Protocol)

A standardized protocol (RFC 4566) providing information about
multimedia initialization settings such as VoIP calls.
RTP (Real-time Transport Protocol):

A standardized transport protocol (RFC 3550) working over UDP at
the transport OSI layer.
RTCP:
A protocol closely linked with RTP (also defined in RFC 3550). It does
not transport any data but gives information about the quality of the
service provided by RTP.
1. SIP 2. SIP 3. 4. 5. 6.
Registration Initialization/Closure SDP RTP RTCP CHECKS
1. SIP Registration
Here is a Wireshark capture of the SIP registration process.

Babar registrates with the trixbox server.
1
6
Let's look at the SIP message headers:
1. REGISTER
The client tries to register with the server.
REGISTER sip:local SIP/2.0

Via: SIP/2.0/UDP 192.168.1.102:15772;branch=z9hG4bK-d87543-
280a581fa364af43-1--d87543-;rport
Max-Forwards: 70
Contact:
To: "Babar"
From: "Babar";tag=11573036
Call-ID: ZGVmYmM0OWRhNzYyMmI5M2FmODIwZjk1YTA2ZTI2Y2I.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0
2. 100 - Trying
The server indicates to the client that it is performing searches.
SIP/2.0 100 Trying

280a581fa364af43-1--d87543-;
received=192.168.1.102;rport=15772
To: "Babar"
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Contact:
Content-Length: 0
3. 401 - Unauthorized
The server rejects the client registration and sends it back a

challenge digest composed of an algorithm type, a "realm" and a
"nonce".
The "nonce" is a random value created on the Asterisk server and
sent to the client. It has a limited lifetime which prevents replay
attacks. Each challenge digest contains a different nonce value.
The "realm" is the SIP domain name.
Digest authentication checks that both communicate parties know a

shared password.
401 Unauthorized
280a581fa364af43-1--d87543-;
received=192.168.1.102;rport=15772
To: "Babar";tag=as1647de36
CSeq: 1 REGISTER
NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="29b8191d"
Content-Length: 0
4. REGISTER
The client sends a new registration request but this time with a
digest response composed of the:
"username", "realm", "nonce", "uri", "response" and the algorithm.
The "URI" (Uniform Resource Identifier) is a string of characters used

to identify a resource.
The "nonce" sent by the server is used to compute a "response".
REGISTER sip:local SIP/2.0

5f795c5af206133a-1--d87543-;rport
Max-Forwards: 70
Contact:
To: "Babar"
CSeq: 2 REGISTER
Expires: 3600
SUBSCRIBE, INFO
Authorization: Digest
username="203",realm="asterisk",nonce="29b8191d",uri="sip
:local",
response="7306cfba1b131f2f04363b68d908f855",algorithm
=MD5
Content-Length: 0
5. 100 - Trying
The server warns the client that it is performing searches.
SIP/2.0 100 Trying

5f795c5af206133a-1--d87543-;
received=192.168.1.102;rport=15772
To: "Babar"
CSeq: 2 REGISTER
NOTIFY
Contact:
Content-Length: 0
6. 200 - OK
After computation, the server is able, to validate the client password

thanks to the digest response it just received.
With the digest authentication process, no password is exchanged
between the client and the server.
The server can send a message to the client to validate the

registration.
SIP/2.0 200 OK
5f795c5af206133a-1--d87543-;
received=192.168.1.102;rport=15772
To: "Babar";tag=as1647de36
CSeq: 2 REGISTER
NOTIFY
Expires: 3600
Contact: ;expires=3600
Date: Fri, 21 Dec 2007 22:15:51 GMT
Content-Length: 0
SIP registration picture
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2. SIP Initialization/Closure & SDP
Here is a Wireshark capture of the SIP initialization and closure

processes.
Bambou (extension 202) calls Babar (extension 203), talks to it and
then hangs up (closure/termination).
Let us see in detail the steps needed for SIP to establish a VoIP call
before voice data can be exchanged between two parties.
The process to establish an SIP link between two hosts is very

similar to the one used for TCP:
TCP SIP
step1
SYN INVITE
:
step2
SYN/ACK 200 - OK
:
step3
ACK ACK
:
1
10
11
12
13
14
The same steps displayed by Wireshark. (click to enlarge)
Let us look at the SIP message headers:
1. INVITE
The client indicates to the server that it wants to establish a phone

call.
INVITE sip:203@192.168.1.222 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.106;rport;branch=z9hG4bKmazrqogm
Max-Forwards: 70
To: < sip:203@192.168.1.222>
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 565 INVITE
Contact: < sip:202@192.168.1.106>
Content-Type: application/sdp
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,IN
FO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
Content-Length: 307
2. 407 - Proxy Authentication Required
The server rejects the client invitation and sends back a challenge
digest composed of an algorithm type, a "realm" and a "nonce".
The "nonce" is a random value created on the Asterisk server and
sent to the client. It has a limited lifetime which prevents replay
attacks. Each challenge digest contains a different nonce value.
Digest authentication checks that both communicating parties know

a shared password.
The "realm" is the SIP domain name.
SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP
192.168.1.106;branch=z9hG4bKmazrqogm;received=192.168.1.106;r
port=5060
To: < sip:203@192.168.1.222>;tag=as219a888b
CSeq: 565 INVITE
NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="138dd154"
Content-Length: 0
3. ACK
The client acknowledges the message
ACK sip:203@192.168.1.222 SIP/2.0

Max-Forwards: 70
To: < sip:203@192.168.1.222>;tag=as219a888b
CSeq: 565 ACK
Content-Length: 0
4. INVITE
The client sends a new invitation request but this time with a digest
response composed of the:
"username", "realm", "nonce", "uri", "response" and the algorithm.
The "nonce" sent by the server is used to compute a "response".

The "URI" (Uniform Resource Identifier) is a string of characters used
to identify a resource.
After computation, the server will be able, to validate the client

password with the digest response it just received.
With the digest authentication process, no password is exchanged
between the client and the server.
INVITE sip:203@192.168.1.222 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.106;rport;branch=z9hG4bKtomragum
Max-Forwards: 70
Proxy-Authorization: Digest
username="202",realm="asterisk",nonce="138dd154",uri="sip
:203@192.168.1.222",
response="c46a93637d0af311c7f9cd3bb542cd08",algorithm
=MD5
To: < sip:203@192.168.1.222>
CSeq: 566 INVITE
Contact: < sip:202@192.168.1.106>
Allow:
FO,MESSAGE
Content-Length: 307
5. 100 - Trying
The server warns the sender (Bambou - 202) that it is trying to reach
the recipient (Babar - 203)
SIP/2.0 100 Trying

Via: SIP/2.0/UDP
192.168.1.106;branch=z9hG4bKtomragum;received=192.168.1.106;r
port=5060
To: < sip:203@192.168.1.222>
CSeq: 566 INVITE
NOTIFY
Contact: < sip:203@192.168.1.222>
Content-Length: 0
SIP Process picture
6. INVITE
The server invites the recipient (Babar - 203).
INVITE sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558
SIP/2.0
Via: SIP/2.0/UDP 192.168.1.222:5060;branch=z9hG4bK407630a5;rport
From: "Bambou" < sip:202@192.168.1.222>;tag=as65501eef

To: < sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558>
Contact: < sip:202@192.168.1.222>
Call-ID: 4ddd4d6e5fc3aacf5e6994da26ac2f94@192.168.1.222
CSeq: 102 INVITE
Max-Forwards: 70
Date: Sun, 16 Dec 2007 20:15:10 GMT
NOTIFY
Content-Length: 240
7. 180 - Ringing
The server sends a message to the sender (Bambou - 202) in order

to ring its phone.
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP
port=5060
To: < sip:203@192.168.1.222>;tag=as0de70729
CSeq: 566 INVITE
NOTIFY
Contact: < sip:203@192.168.1.222>
Content-Length: 0
8. 180 - Ringing
The recipient (Babar - 203) tells the server that its phone is ringing.
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP
192.168.1.222:5060;branch=z9hG4bK407630a5;rport=5060
Contact: <
sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558>
To: <
sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558>;tag=115
eda75
From: "Bambou"< sip:202@192.168.1.222>;tag=as65501eef
CSeq: 102 INVITE
Content-Length: 0
9. 200 - OK
The recipient (Babar - 203) confirms the server invitation.
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.222:5060;branch=z9hG4bK407630a5;rport=5060
Contact: <
To: <
eda75
CSeq: 102 INVITE
SUBSCRIBE, INFO
Content-Length: 187
SIP Process picture
10. ACK
The server acknowledges the recipient (Babar - 203) confirmation.
ACK sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558
SIP/2.0
Via: SIP/2.0/UDP 192.168.1.222:5060;branch=z9hG4bK64b6b476;rport

To: <
eda75
Contact: < sip:202@192.168.1.222>
CSeq: 102 ACK
Max-Forwards: 70
Content-Length: 0
11. 200 - OK
The server confirms the sender (Bambou - 202) invitation (step 4).
SIP/2.0 200 OK
Via: SIP/2.0/UDP
port=5060
To: < sip:203@192.168.1.222>;tag=as0de70729
CSeq: 566 INVITE
NOTIFY
Contact: < sip:203@192.168.1.222>
Content-Length: 240
12. ACK
The sender (Bambou - 202) acknowledges the server confirmation.

The phone call can begin. The RTP protocol will transport the VoIP
packets and the RTCP will control the link quality.
ACK sip:203@192.168.1.222 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.106;rport;branch=z9hG4bKixdtxpvy
Max-Forwards: 70
Proxy-Authorization: Digest
username="202",realm="asterisk",nonce="138dd154",uri="sip:203@
192.168.1.222",
response="c46a93637d0af311c7f9cd3bb542cd08",algorithm=MD5
To: < sip:203@192.168.1.222>;tag=as0de70729
CSeq: 566 ACK
Content-Length: 0
13. BYE
The sender (Bambou - 202) hangs up.
BYE sip:203@192.168.1.222 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.106;rport;branch=z9hG4bKassptnfl
Max-Forwards: 70
To: < sip:203@192.168.1.222>;tag=as0de70729
CSeq: 567 BYE
Content-Length: 0
SIP Process picture
14. 200 - OK
The server confirms the sender's (Bambou - 202) BYE message with
an OK.
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.106;branch=z9hG4bKassptnfl;received=192.168.1.106;rpor
t=5060
To: < sip:203@192.168.1.222>;tag=as0de70729
CSeq: 567 BYE
NOTIFY
Contact: < sip:203@192.168.1.222>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
15. BYE
The server tells the recipient (Babar - 203) that the sender (Bambou
- 202) hanged up.
BYE sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558
SIP/2.0
Via: SIP/2.0/UDP 192.168.1.222:5060;branch=z9hG4bK07b32129;rport

To: <
eda75
CSeq: 103 BYE
Max-Forwards: 70
Content-Length: 0
16. 200 - OK
The recipient (Babar - 203) confirms the BYE message with an OK.
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.222:5060;branch=z9hG4bK07b32129;rport=5060
Contact: <
To: <
eda75
CSeq: 103 BYE
Content-Length: 0
SIP Process picture
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3. SDP
SDP is closely related to SIP.

The Wireshark capture is the same one used to illustrate the SIP
initialization and closure processes.
Below, the SIP message header of the first capture line (INVITE)
containing SDP infomation (in bold).
INVITE sip:203@192.168.1.222 SIP/2.0

Max-Forwards: 70
To: < sip:203@192.168.1.222>
CSeq: 565 INVITE
Contact: < sip:202@192.168.1.106>
Allow:
FO,MESSAGE
Content-Length: 307
v=0
o=201 2086878285 1015399812 IN IP4 192.168.1.106
s=-
c=IN IP4 192.168.1.106
t=0 0
m=audio 8000 RTP/AVP 98 97 8 0 3 101
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
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4. RTP
The RTP protocol is used to transport the voice data and thus enable
to two people to speak together.
Top of the page
5. RTCP
RTCP is used in conjunction with RTP to check the communication

quality.
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6. CHECKS
Extended statistics can be obtained in the "reports" sections of the

FreePBX interface.
Call logs:
Daily load:
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SIP Call Illustration

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SIP Call Illustration

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The protocols used in an IP phone call are:

SIP (Session Initiation Protocol):

SDP (Session Description Protocol)

RTP (Real-time Transport Protocol):

Here is a Wireshark capture of the SIP registration process.

The client tries to register with the server.

REGISTER sip:local SIP/2.0

The server indicates to the client that it is performing searches.

SIP/2.0 100 Trying

The server rejects the client registration and sends it back a

Digest authentication checks that both communicate parties know a

The "URI" (Uniform Resource Identifier) is a string of characters used

The "nonce" sent by the server is used to compute a "response".

REGISTER sip:local SIP/2.0

The server warns the client that it is performing searches.

SIP/2.0 100 Trying

After computation, the server is able, to validate the client password

The server can send a message to the client to validate the

Top of the page

2. SIP Initialization/Closure & SDP

Here is a Wireshark capture of the SIP initialization and closure

The process to establish an SIP link between two hosts is very

Let us look at the SIP message headers:

The client indicates to the server that it wants to establish a phone

INVITE sip:203@192.168.1.222 SIP/2.0

Digest authentication checks that both communicating parties know

SIP/2.0 407 Proxy Authentication Required

The client acknowledges the message

ACK sip:203@192.168.1.222 SIP/2.0

The "nonce" sent by the server is used to compute a "response".

After computation, the server will be able, to validate the client

INVITE sip:203@192.168.1.222 SIP/2.0

SIP/2.0 100 Trying

The server invites the recipient (Babar - 203).

From: "Bambou" < sip:202@192.168.1.222>;tag=as65501eef

The server sends a message to the sender (Bambou - 202) in order

SIP/2.0 180 Ringing

SIP/2.0 180 Ringing

The recipient (Babar - 203) confirms the server invitation.

The server acknowledges the recipient (Babar - 203) confirmation.

From: "Bambou" < sip:202@192.168.1.222>;tag=as65501eef

The sender (Bambou - 202) acknowledges the server confirmation.

ACK sip:203@192.168.1.222 SIP/2.0

The sender (Bambou - 202) hangs up.

BYE sip:203@192.168.1.222 SIP/2.0

From: "Bambou" < sip:202@192.168.1.222>;tag=as65501eef

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SDP is closely related to SIP.

INVITE sip:203@192.168.1.222 SIP/2.0

Top of the page

RTCP is used in conjunction with RTP to check the communication

Top of the page

Extended statistics can be obtained in the "reports" sections of the

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