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Trixbox - The Easy Tutorial - Analyzes

Trixbox Analyzes
Last Change : Dec 07 2010 french flagenglish flag


Tool
Install
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Details What is Trixbox ?
Screenshots
Prerequisites
Configurations
Softphones
---- Statistics ----
Voice protocols
Link quality




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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 Registration 2. SIP Initialization/Closure 3. SDP 4. RTP 5. RTCP 6. CHECKS



1. SIP Registration

Here is a Wireshark capture of the SIP registration process.
Babar registrates with the trixbox server.

wireshark capture sip registration










1

2


3


4

5

6

sip registration process register ok 401 180 trying
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
Via: SIP/2.0/UDP 192.168.1.102:15772;branch=z9hG4bK-d87543-280a581fa364af43-1--d87543-;
   received=192.168.1.102;rport=15772
From: "Babar";tag=11573036
To: "Babar"
Call-ID: ZGVmYmM0OWRhNzYyMmI5M2FmODIwZjk1YTA2ZTI2Y2I.
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
Via: SIP/2.0/UDP 192.168.1.102:15772;branch=z9hG4bK-d87543-280a581fa364af43-1--d87543-;
   received=192.168.1.102;rport=15772
From: "Babar";tag=11573036
To: "Babar";tag=as1647de36
Call-ID: ZGVmYmM0OWRhNzYyMmI5M2FmODIwZjk1YTA2ZTI2Y2I.
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 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
Via: SIP/2.0/UDP 192.168.1.102:15772;branch=z9hG4bK-d87543-5f795c5af206133a-1--d87543-;rport
Max-Forwards: 70
Contact:
To: "Babar"
From: "Babar";tag=11573036
Call-ID: ZGVmYmM0OWRhNzYyMmI5M2FmODIwZjk1YTA2ZTI2Y2I.
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
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
Via: SIP/2.0/UDP 192.168.1.102:15772;branch=z9hG4bK-d87543-5f795c5af206133a-1--d87543-;
   received=192.168.1.102;rport=15772
From: "Babar";tag=11573036
To: "Babar"
Call-ID: ZGVmYmM0OWRhNzYyMmI5M2FmODIwZjk1YTA2ZTI2Y2I.
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 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
Via: SIP/2.0/UDP 192.168.1.102:15772;branch=z9hG4bK-d87543-5f795c5af206133a-1--d87543-;
   received=192.168.1.102;rport=15772
From: "Babar";tag=11573036
To: "Babar";tag=as1647de36
Call-ID: ZGVmYmM0OWRhNzYyMmI5M2FmODIwZjk1YTA2ZTI2Y2I.
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 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).

sip steps ack invite ringing OK bye 407 180 200 wireshark capture
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


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3

4

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6

7

8


9

10

11


12














13

14

15

16
sip steps ack invite ringing OK bye 407 180 200
The same steps displayed by Wireshark. (click to enlarge)

wireshark capture sip wireshark
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,INFO,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;rport=5060
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
To: < sip:203@192.168.1.222>;tag=as219a888b
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 565 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 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
Via: SIP/2.0/UDP 192.168.1.106;rport;branch=z9hG4bKmazrqogm
Max-Forwards: 70
To: < sip:203@192.168.1.222>;tag=as219a888b
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 565 ACK
User-Agent: Twinkle/1.1
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>
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 566 INVITE
Contact: < sip:202@192.168.1.106>
Content-Type: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
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;rport=5060
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
To: < sip:203@192.168.1.222>
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 566 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 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
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 16 Dec 2007 20:15:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
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 192.168.1.106;branch=z9hG4bKtomragum;received=192.168.1.106;rport=5060
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
To: < sip:203@192.168.1.222>;tag=as0de70729
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 566 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 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=115eda75
From: "Bambou"< sip:202@192.168.1.222>;tag=as65501eef
Call-ID: 4ddd4d6e5fc3aacf5e6994da26ac2f94@192.168.1.222
CSeq: 102 INVITE
User-Agent: X-Lite release 1011s stamp 41150
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: < sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558>
To: < sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558>;tag=115eda75
From: "Bambou"< sip:202@192.168.1.222>;tag=as65501eef
Call-ID: 4ddd4d6e5fc3aacf5e6994da26ac2f94@192.168.1.222
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
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
From: "Bambou" < sip:202@192.168.1.222>;tag=as65501eef
To: < sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558>;tag=115eda75
Contact: < sip:202@192.168.1.222>
Call-ID: 4ddd4d6e5fc3aacf5e6994da26ac2f94@192.168.1.222
CSeq: 102 ACK
User-Agent: Asterisk PBX
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 192.168.1.106;branch=z9hG4bKtomragum;received=192.168.1.106;rport=5060
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
To: < sip:203@192.168.1.222>;tag=as0de70729
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 566 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: < sip:203@192.168.1.222>
Content-Type: application/sdp
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
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 566 ACK
User-Agent: Twinkle/1.1
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
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 567 BYE
User-Agent: Twinkle/1.1
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;rport=5060
From: "Bambou" < sip:202@192.168.1.222>;tag=kbcql
To: < sip:203@192.168.1.222>;tag=as0de70729
Call-ID: jcaqhewsybtyksb@192.168.1.106
CSeq: 567 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 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
From: "Bambou" < sip:202@192.168.1.222>;tag=as65501eef
To: < sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558>;tag=115eda75
Call-ID: 4ddd4d6e5fc3aacf5e6994da26ac2f94@192.168.1.222
CSeq: 103 BYE
User-Agent: Asterisk PBX
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: < sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558>
To: < sip:203@192.168.1.102:9097;rinstance=fc31ac7abb1cc558>;tag=115eda75
From: "Bambou"< sip:202@192.168.1.222>;tag=as65501eef
Call-ID: 4ddd4d6e5fc3aacf5e6994da26ac2f94@192.168.1.222
CSeq: 103 BYE
User-Agent: X-Lite release 1011s stamp 41150
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.

wireshark capture sip registration
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
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,INFO,MESSAGE
Supported: replaces,norefersub,100rel
User-Agent: Twinkle/1.1
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.

rtp protocol voip voice wireshark capture
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5. RTCP

RTCP is used in conjunction with RTP to check the communication quality.

rtcp protocol voip voice wireshark capture
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6. CHECKS

Extended statistics can be obtained in the "reports" sections of the FreePBX interface.

Call logs:

asterisk freepbx reports call logs voip voice
Daily load:

asterisk freepbx reports daily load voip voice
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