Setup VoIP System and Interconnection with

LTE networkMohammad Nazmul Hossain

Md. Farhad HossainTowfique Imam Chowdhury

AbstractSetting 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)

G.711 payload (20 ms)

bits

bits

gsm codec bandwidth (35 kbps)

gsm payload (20 ms)

bits

bits

H.263 bandwidth (220 kbps)

H.263 payload (70 ms)

bits

bits

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)

• 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).

bits

Video (84 kbps) vs Audio (230 kbps) stream

Video stream

Audio streambits

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‘ comparisonSession Payload type Packets lost Packet loss % Mean Jitter (ms)

1G711A 28 0.9 2.42

H263 25 1.4 4.53

2G711A 26 0.9 0.20

H263 21 1.2 0.80

3G711A 34 1.1 2.46

H263 17 1.0 3.80

4G711A 33 1.0 1.75

H263 8 0.4 2.55

5G711A 29 0.9 3.52

H263 59 3.2 5.41

6G711A 33 1.1 1.94

H263 9 0.5 3.04

G711A vs H263 Packet loss

G711A Packet Loss H263 Packet Loss0

10

20

30

40

50

60

70

Session 1 Session 2 Session 3 Session 4 Session 5 Session 6

G711A vs H263 Mean Jitter

G711A Mean Jitter H263 Mean Jitter0

1

2

3

4

5

6

Session 1 Session 2 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 Ekiga‘s 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.confRTP configuration file (rtp.conf)

;; RTP Configuration;[general];; RTP start and RTP end configure start and end addresses;rtpstart=25008rtpend=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 it’s 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