Setup VoIP System and Interconnection with LTE network
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The document outlines the setup of a VoIP system interconnected with an LTE network using a communication framework, detailing necessary hardware and software resources. It provides information on various codecs, their bandwidth, and payload comparisons alongside a discussion on potential call issues such as audio/video loss. Additionally, it includes technical configurations for SIP and RTP protocols essential for establishing and managing VoIP calls.
![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](https://image.slidesharecdn.com/ngnpresentation-170309100904/85/Setup-VoIP-System-and-Interconnection-with-LTE-network-24-320.jpg)
Setup VoIP System and Interconnection with LTE network
- 1. Setup VoIP System and Interconnection with LTE network Mohammad Nazmul Hossain Md. Farhad Hossain Towfique Imam Chowdhury
- 2. 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.
- 4. 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.
- 5. UDP header is smaller than TCP header UDP Header TCP Header
- 6. SIP (Session Initiation Protocol)
- 9. Total RTP vs one voice stream
- 10. Source to Destination voice stream
- 11. Voice payload
- 12. Jitter
- 13. G.711 codec bandwidth (85kbps) G.711 payload (20 ms) bits bits
- 14. gsm codec bandwidth (35 kbps) gsm payload (20 ms) bits bits
- 15. H.263 bandwidth (220 kbps) H.263 payload (70 ms) bits bits
- 16. 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
- 17. Video vs Audio stream Video stream Audio stream bits
- 18. RTP, Video & Audio stream comparison Total RTP Video Stream Source video stream Voice stream bits
- 20. Session Payload type Packets lost Packet loss % Mean Jitter (ms) 1 G711A 28 0.9 2.42 H263 25 1.4 4.53 2 G711A 26 0.9 0.20 H263 21 1.2 0.80 3 G711A 34 1.1 2.46 H263 17 1.0 3.80 4 G711A 33 1.0 1.75 H263 8 0.4 2.55 5 G711A 29 0.9 3.52 H263 59 3.2 5.41 6 G711A 33 1.1 1.94 H263 9 0.5 3.04
- 21. G711A vs H263 Packet loss 0 10 20 30 40 50 60 70 G711A Packet Loss H263 Packet Loss Session 1 Session 2 Session 3 Session 4 Session 5 Session 6
- 22. G711A vs H263 Mean Jitter 0 1 2 3 4 5 6 G711A Mean Jitter H263 Mean Jitter Session 1 Session 2 Session 3 Session 4 Sesion 5 Session 6
- 23. 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
- 24. 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
- 25. 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.
- 26. 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
- 27. 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