voip system and interconnection with lte network
DESCRIPTION
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.TRANSCRIPT
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