Setup VoIP System and

Interconnection with LTE

network
Mohammad Nazmul Hossain
Md. Farhad Hossain
Towfique Imam Chowdhury

Abstract
Setting up VoIP management server using a
communication framework and let the users
from LTE networks to register and make
voice calls over IP system as well as video
session.

The Project Architecture

Resources:
A PC with Ubuntu OS to install open source Asterisk server.
A 2nd PC to install the softphone Zoiper and Ekiga client installed
(Ubuntu OS).
A webcam.
A headphone.
Two IP Phones (Grandstream GXV3140 & snom 360).
Three smartphones with Antisip app installed as a VoIP client.
Smartphones also have Cisco Any Connect software installed for VPN
connection.
The server pc also have a zoiper client.

UDP header is smaller than TCP

header

UDP Header

TCP Header

SIP (Session Initiation Protocol)

Session Description
Protocol (SDP)

Call setup Process

Total RTP vs one voice stream

Source to Destination voice stream

Voice payload

Jitter

G.711 codec bandwidth (84kbps)

bit
s

G.711 payload (20 ms)

bit
s

gsm codec bandwidth (35 kbps)

bits

gsm payload (20 ms)

bits

H.263 bandwidth (220 kbps)

bit
s

H.263 payload (70 ms)

bit
s

Comparison of bit rate & payload for

different codecs

Codec

Bit Rate

Payload
(ms)

G.711

84 kbps

20 ms

gsm

35 kbps

20 ms

G.722

86 kbps

20 ms

H.263

220 kbps

70 ms

H.264

230 kbps

70 ms

G.711 codec bitrate (85)

bits

Nominal bitrate for G.711 is 64 kbps.

But we have found 84 kbps.
VoIP packet = (VoIP header + voice payload).
Physical network VoIP packet = Network interface
headers + (VoIP header + voice payload).

Video (84 kbps) vs Audio (230 kbps)

stream
Video
stream
Audio
stream

bits

RTP, Video & Audio stream

comparison
Total RTP (1180 kbps)
Total Video Stream (850
kbps)
Source video stream (230
kbps)
Voice stream (84
kbps)

bits

A call session measurement for 60

seconds

6 call sessions comparison

Session
1
2
3
4
5
6

Payload
type

Packets
lost

Packet loss
%

Mean Jitter
(ms)

G711A
H263
G711A
H263
G711A
H263
G711A
H263
G711A
H263
G711A
H263

28
25
26
21
34
17
33
8
29
59
33
9

0.9
1.4
0.9
1.2
1.1
1.0
1.0
0.4
0.9
3.2
1.1
0.5

2.42
4.53
0.20
0.80
2.46
3.80
1.75
2.55
3.52
5.41
1.94
3.04

G711A vs H263 Packet loss

70
60
50
40
30
20
10
0

G711A Packet Loss

Session 1

Session 2

H263 Packet Loss

Session 3

Session 4

Session 5

Session 6

G711A vs H263 Mean Jitter

6
5
4
3
2
1
0

G711A Mean Jitter

Session 1

Session 2

H263 Mean Jitter

Session 3

Session 4

Sesion 5

Session 6

Ekiga soft client use port 5060 !

Ekiga is a softphone which we have used for video call
in the Ubuntu OS.
But Ekigas default port no. is 5060.
Bindport=5061
sudo netstat -t -u -l -n --program | grep 5060
This command will show the certain port is listening to
which application.
SIP from 5000 to 5100

rtp.conf

RTP configuration file (rtp.conf)

;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=25008
rtpend=25025
But our port
range is
25008 25027

rtp.conf
First port number must be even number. (25008)
Last port number must be defined an odd number.
(25025)
Asterisk will automatically use the next even number for
its last port range.
For example if rtpend=25027 (last port range) then
Asterisk will use 25028 as its last port number.

No video / No audio
Sometimes we had problem that call connected but no
audio or no video.
Both parties must have the same voice & video codec
enabled.

allow=alaw
allow=ulaw
allow=h263

SIP not loaded in Asterisk CLI>

No such command sip show peers.
/etc./asterisk/modules.conf
noload => chan_sip.so
load => chan_sip.so
preload => chan_sip.so