SIP Overview

VoLTE Planning and Optimization [FL16A]

Objectives

After completing this module, the participant will be able to:

• Describe Session Initiation Protocol SIP
• Describe Session Description Protocol SDP
• Describe Real Time Transport Protocol RTP
VoIP Signaling and Bearer Flows
VoIP Bearer Packets
Session Initiation Protocol (SIP)
SIP Uniform Resource Identifiers (URIs)

sip:host@domain
sip:host.domain

SIP URI Examples

sip:ken.c@voip.net

sip:johnm@motorola.com

sips:scott.p@itsp.net

sip:412-339-
1882@sprint.net
sip:marys@195.23.18.99

sip:server1.itsp.net
SIP Requests
SIP Messages
The basic SIP messages, described in RFC 3261
Method Description
INVITE Sets up or changes a multimedia call.

Confirms the final response for INVITE; closes the

ACK
INVITE process.

BYE Releases a multimedia call.

CANCEL Cancels a pending request
OPTIONS Signals supported features.
REGISTER Registers with a SIP location server.
Extended SIP messages
Method Source Description
SUBSCRIBE RFC 3265 Subscribes to a service.
Notifies a subscribed user that an event has
NOTIFY RFC 3265
occurred.
Carries a non-SIP message transparently through the
INFO RFC 2976
SIP network.
PRACK RFC 3262 Acknowledges 1XX provisional responses.
MESSAGE RFC 3428 Supports “instant messages.”
REFER RFC 3515 Refers to an earlier INVITE for Call Forwarding, etc.
Confirms resource reservation (QoS) preconditions
UPDATE RFCs 3311 & 3312
have been met.
SIP Responses
Response Codes

Response Codes Commonly Used SIP Responses

Response
Codes Type Description
Response Code Default Text
Informational – typically used with INVITE
1XX Provisional
requests
2XX Final Success 100 Trying
3XX Final Redirection
4XX Final Failure of request 180 Ringing

5XX Final Server failure 183 Session Progress

6XX Final Global failure
200 OK

202 Accepted

301 Moved Permanently

302 Moved Temporarily

401 Unauthorized

408 Request Timeout

415 Unsupported Media Type

SIP Transactions

• A transaction consists of a Request and all of its

provisional (1XX) and final (2XX-6XX) Responses
• Every Response carries the same Command
Sequence (CSeq) number as the original Request
SIP Error Recovery

• The SIP endpoint expects a final Response within a timeout period (default
is 500 milliseconds)
• If a Response is not received within the timeout period, the Request is
retransmitted
SIP Headers
SIP Calls and Dialogs

•Call identified by From, To, and Call-ID headers (1 call)

•Dialog identified by From+Tag, To+Tag, and Call-ID
Multipoint Call
Session Description Protocol (SDP)

• SIP creates and maintains the call relationship

• SDP establishes the call details
• SDP is described in RFCs 4566 and 3264
• SDP is carried in the SIP message body

SDP Models
• SDP Announcement
• SDP Offer/Answer
• SDP Declaration/Offer/Answer
SDP Offer/Answer Model
SDP Headers

v=0
o=- IN IP4 11.22.1.76
s=- Session Description
c=IN IP4 11.22.1.76
t= 0 0
m=audio 11088 RTP/AVP 18 3
b=AS:8
Media 1 Description
a=rtpmap:18 G.729
a=rtpmap:3 GSM
m=video 12000 RTP/AVP 31
a=rtpmap:31 H.261 Media 2 Description
a=sendonly

SDP Header Header Name

Description
Version v= The SDP version. The current SDP version is 0.
The originating user ID, SDP session ID, offer version, and user IP address. A
Origin o= hyphen indicates some user information is contained in a SIP (From) header.
An SDP declaration will use IP address 0.0.0.0.
Subject s= The subject of this session – usually set to a hyphen.
Contact c= The contact user ID or IP address.
Session start and end time. Normally set to t= 0 0, allowing SIP to control
Time t=
session start and end time.
SDP Media Description
RTP/AVP Profiles

AVP AVP Name Description

0 PCMU 64 kbps Pulse Code Modulation, mu-law
3 GSM 13 kbps GSM audio encoding
8 PCMA 64 kbps Pulse Code Modulation, A-law
18 G.729 8 kbps CS-ACELP audio encoding
31 H.261 N x 64 kbps video encoding
32 MPEG1 or MPEG2 encoding, visual
MPV
stream
33 MP2T MPEG2 Transport (audio and video)

RTP Mapping Example

m= audio 8498 RTP/AVP 97 0 18

a= rtpmap: 97 evrc/8000

a= rtpmap: 0 pcmu/8000

a= rtpmap: 18 g729/8000
Media Attributes

SDP Media Attributes Example

m=video 9044 RTP/AVP 31 33

b=AS:128
a=rtpmap: 31 h.261
a=rtpmap: 33 mp2t
a=sendonly
 UE MME/SAE/PGW S-CSCF
Registering with the SIP Server
Attach, Default Bearer setup, NAS Authentication, Integrity
Protection 1
Activate EPS Bearer (QCI5) and DRB

Activate EPS Bearer Accept (QCI5) and DRB

Store P-CSCF IP
Address

Extract user public

identity from ISIM

Allocate client and

server ports

2
REGISTER

Registration procedures with

HSS

401 Unauthorized:
3
Verify AUTN & calculate
RES

4 REGISTER

Download service control

information from HSS

5 200 OK,
Call Setup
VoLTE Call Flow – NSN Smart Lab
Precondition Mechanism
Originating UE IMS Network Terminating UE

Initiate call

INVITE

100 Trying
OPTIONS

200 OK

INVITE

100 Trying

183 Session Progress

EPS Bearer Activation for QCI1 and Audio Video

183 Session Progress

5 Path Setup

PRACK

EPS Bearer Activation for QCI1 and Audio Video

6
Path Setup

PRACK

200 OK 200 OK

UPDATE 7 UPDATE

200 OK 8 200 OK

Ringing
180 Ringing 9 180 Ringing
Answer
200 OK 200 OK

ACK
9 ACK

Voice or Video Session

24 RA47072EN16GLA0 © Nokia 2016
VoLTE Call Flow – xx Network
Originating IMS Network Terminating
UE UE
No Precondition Mechanism Initiate
call

INVITE
The VoLTE call setup flow from XX network is illustrated
on the right with the following remarks: 100 Trying

183 Session Progress

1. After 183 Session Progress message IMS signaling INVITE

(QCI 5) bearer can carry RTP audio packets, e.g.

100 Trying
voice announcement or music, to notify originating
UE. Therefore, „180 Ringing‟ message is NOT
required to be forwarded to originating UE.
1 180 Ringing
Ringi
ng

Answ
2. Once MT UE answers the call EPS bearer for QCI1 is 200 OK er

activated after a terminating UE has received ACK

ACK
from IMS, i.e. three-way handshake (INVITE/200
OK/ACK) completes SIP session establishment at MT 2 EPS Bearer Activation for QCI1 and Audio
Video Path Setup
UE.
3. EPS bearer for QCI1 is activated at MO UE once a 200 OK
originating UE has sent ACK and SIP session is then ACK
established - voice communication starts.
3 EPS Bearer Activation for QCI1 and Audio
Video Path Setup

Voice or Video Session

25 RA47072EN16GLA0 © Nokia 2016

Ending a SIP Call
RTP Protocol Stack
RTP Header
RTP Operation
Jitter
Exercise 1 -SIP
Using the captured SIP packets at the end of this exercise, place the five SIP
messages into a call flow. Remember, SIP responses show the same From and
To headers as the original request!

Msg Phone 1 SIP Server Phone 2

1

How did you figure out the call flow?

Exercise 2 – Identifying SIP URIs

Use the captured SIP messages to identify the IP addresses, telephone

numbers, and SIP Uniform Resource Identifiers (URIs) for the two SIP
telephones.
SIP Telephone 1

Telephone
Number
SIP URI

IP Address

SIP Telephone 2

Telephone
Number
Where did you find this information? SIP URI

IP Address
Exercise 3 Interpreting SIP Headers
A. What combination of headers uniquely identifies this SIP call?

B. What Layer 4 protocol and port number were used for the SIP signaling messages? Where did
you find that information?

C. What combination of fields identifies a dialog within this call?

D. What field identifies a transaction within the dialog?

Exercise 4 Interpreting SDP Parameters
The captured SIP packets contain an SDP offer/answer handshake. Interpret the SDP parameters
and answer the following questions.

A. Which message carried the SDP offer? Who sent the message?

B. What voice encoders were listed in the SDP offer? Which vocoder was preferred?

C. Which SIP message carried the SDP answer? What vocoder was selected in the SDP answer?

D. What UDP port numbers were selected for the media flows?
Exercise 5 – Interpreting IP, UDP & RTP Headers

A. What IP addresses are used? Who sent each media packet, Phone 1 or 2? (Hint: refer
to the IP addresses you discovered in the SIP signaling.)

B. What IP quality of service was requested? Hint: check the IP headers.

C. What UDP port numbers are used? Is this consistent with the SDP parameters in
Exercise4?

D. What type of “data” is carried in each RTP packet?

Captured SIP Packets

•SIP Packet 1 of 5 SIP Packet 2 of 5

INVITE sip:202-682-0167@voip.net:5060 SIP/2.0 SIP/2.0 100 Trying
Via: SIP/2.0/UDP sip:512-378-1231@itsp.net:5060;branch=z9hG4bK_1102 Via: SIP/2.0/UDP sip:512-378-
From: "Phone 1"<sip:512-378- 1231@itsp.net:5060;branch=z9hG4bK_1102;received=11.22.3.1
1231@itsp.net;user=phone>;tag=1_1102_f4726 From: "Phone 1"<sip:512-378-1231@itsp.net;user=phone>;tag=1_1102_f4726
To: <sip: 202-682-0167@voip.net;user=phone> To: <sip: 202-682-0167@voip.net;user=phone>
Call-ID: 1851017346@itsp.net Call-ID: 1851017346@itsp.net
CSeq: 1 INVITE CSeq: 1 INVITE
Max-Forwards: 70 Contact: <sip:202-682-0167@180.13.2.3:5060>
Contact: <sip:512-378-1231@itsp.net:5060> Content-Length: 0
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY
,REFER,UPDATE
Content-Length: 429
Content-Type: application/sdp
SIP Packet 3 of 5
v=0
SIP/2.0 180 Ringing
o=- IN IP4 11.22.1.76
Via: SIP/2.0/UDP sip:512-378-
s=-
1231@itsp.net:5060;branch=z9hG4bK_1102;received=11.22.3.1
c=IN IP4 11.22.1.76
From: "Phone 1"<sip:512-378-
t=0 0
1231@itsp.net;user=phone>;tag=1_1102_f4726
m=audio 11088 RTP/AVP 0 8 99 101 102
To: <sip:202-682-
a=rtpmap:99 G.729a/8000
0167@voip.net;user=phone>;tag=000ded61654900
a=rtpmap:101 G.729b/8000
Call-ID: 1851017346@itsp.net
a=rtpmap:102 G.726-24/8000
CSeq: 1 INVITE
Contact: <sip:202-682-0167@180.13.2.3:5060>
Content-Length: 0
 Captured SIP Packets

SIP Packet 4 of 5
SIP/2.0 200 OK SIP Packet 5 of 5
Via: SIP/2.0/UDP sip:512-378- ACK sip:202-682-0167@180.13.2.3:5060 SIP/2.0
1231@itsp.net:5060;branch=z9hG4bK_11 Via: SIP/2.0/UDP sip:512-378-
02;received=11.22.3.1 1231@itsp.net:5060;branch=z9hG4bK_1266
From: "Phone 1"<sip:512-378- From: "Phone 1"<sip:512-378-
1231@itsp.net;user=phone>;tag=1_1102_ 1231@itsp.net;user=phone>;tag=1_1102_f4726
f4726 To: <sip:202-682-
To: <sip:202-682- 0167@voip.net;user=phone>;tag=000ded61654900
0167@voip.net;user=phone>;tag=000ded Call-ID: 1851017346@itsp.net
61654900 CSeq: 1 ACK
Call-ID: 1851017346@itsp.net Max-Forwards: 70
CSeq: 1 INVITE Content-Length: 0
Contact: <sip:202-682-
0167@180.13.2.3:5060>
Content-Type: application/sdp
Content-Length: 140
v=0
o=- IN IP4 180.13.2.3
s=SIP Call
c=IN IP4 180.13.2.3
t=0 0
m=audio 16384 RTP/AVP 0
a=rtpmap:0 PCMU/8000
Captured RTP Bearer Packets
RTP Bearer Packet 1 of 3
Ethernet:
MAC Addresses: Destination = 00:D0:58:72:DB:80, Source = 00:02:B9:B5:8F:20
Protocol: 0x0800 IP
CRC: (Good)
IP Header:
Version: 4
IP Header Length = 20 Bytes
Type of Service:
011 . . . . . : Precedence = Flash
. . .0 . . . . : Delay = Normal
. . . .0 . . . : Throughput = Normal
. . . . .0 . . : Reliability = Normal
. . . . . .0 . : Cost = Normal
. . . . . . . 0: Reserved
Total IP Datagram length: 200 bytes
ID: 0x22D5
Fragment: Not Fragmented
Time to live: 254
Protocol: 17 (UDP)
Header checksum: 0x7FBE (Good)
IP Addresses: Source = 11.22.1.76 Destination = 180.13.2.3
UDP Header:
Ports: Source= 11088. Destination = 16384
UDP Datagram length: 180 bytes
Checksum: 0x0000
RTP Header:
Version: 2, Padding: 0, Extension: 0, CSRC Count: 0, Marker: 0,
Payload Type: PCMU(G.711) (0), Audio/Video: Audio, Clock Rate (Hz): 8000
Sequence Number: 1400, Timestamp: 1183210951, SSRC: 34537731
RTP Data:
Length = 160 bytes
0000 FC FD 72 79 F7 F8 77 77 7C FE F9 F9 7C 70 7A F1
0010 F5 72 79 7E 7C
Captured RTP Bearer Packets
RTP Bearer Packet 2 of 3
Ethernet:
MAC Addresses; Destination = 00:02:B9:B5:8F:20, Source = 00:D0:58:72:DB:80
Protocol: 0x0800 IP
CRC: (Good)
IP Header:
Version: 4
IP Header Length: 20 Bytes
Type of Service:
011 . . . . . : Precedence = Flash
. . .0 . . . . : Delay = Normal
. . . .0 . . . : Throughput = Normal
. . . . .0 . . : Reliability = Normal
. . . . . .0 . : Cost = Normal
. . . . . . . 0: Reserved
Total IP length: 200 bytes
ID: 0xB658
Fragment: Not Fragmented
Time to live: 252
Protocol: 17 (UDP)
Header checksum: 0xEE3A (Good)
IP Addresses: Source = 180.13.2.3, Destination = 11.22.1.76
UDP Header:
Ports: Source = 16384, Destination = 11088
UDP length: 180 bytes
Checksum: 0x0000
RTP Header:
Version: 2, Padding: 0, Extension: 0, CSRC Count: 0, Marker: 0,
Payload Type: PCMU(G.711) (0), Audio/Video: Audio, Clock Rate (Hz): 8000
Sequence Number: 5991, Timestamp: 3710524699, SSRC: 535953923
RTP Data:
Length = 160 bytes
0000 70 7A 76 7E 7E FE FC F9 F1 FC FA FA FC F8 F7 FA
0010 FA FB F6 F7
Captured RTP Bearer Packets
RTP Bearer Packet 3 of 3
Ethernet:
MAC Addresses: Destination = 00:D0:58:72:DB:80, Source = 00:02:B9:B5:8F:20
Protocol: 0x0800 IP
CRC: (Good)
IP Header:
Version: 4
IP Header Length: 20 Bytes
Type of Service:
011 . . . . . : Precedence = Flash
. . .0 . . . . : Delay = Normal
. . . .0 . . . : Throughput = Normal
. . . . .0 . . : Reliability = Normal
. . . . . .0 . : Cost = Normal
. . . . . . . 0: Reserved
Total IP length: 200 bytes
ID: 0x22D5
Fragment: Not Fragmented
Time to live: 254
Protocol: 17 (UDP)
Header checksum: 0x7FBE (Good)
IP Addresses: Source = 11.22.1.76, Destination = 180.13.2.3
UDP Header:
Ports: Source = 11088, Destination = 16384
UDP length: 180 bytes
Checksum: 0x0000
RTP Header:
Version: 2, Padding: 0, Extension: 0, CSRC Count: 0, Marker: 0,
Payload Type: PCMU(G.711) (0), Audio/Video: Audio, Clock Rate (Hz): 8000
Sequence Number: 1401, Timestamp: 1183211111, SSRC: 34537731
RTP Data:
Length = 160 bytes
0000 FF 7C 7E FD FC 7D 7B 7C FE FD 7B 7A 7C FE FB FE
0010 7C 79 FD F8 7E
Module Review
This module:
• Described Session Initiation Protocol SIP
• Described Session Description Protocol SDP
• Described Real Time Transport Protocol RTP