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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.
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
TCP Header
Session Description
Protocol (SDP)
Voice payload
Jitter
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
bits
bits
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
Session 2
Session 4
Session 5
Session 6
Session 2
Session 4
Sesion 5
Session 6
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