Scribd
Upload
331 views15 pages

VoIP in RouterOS PDF

This document provides an overview of mastering VoIP in RouterOS. It discusses codecs like G.711 and G.729 used for VoIP, protocol requirements around packet loss, latency and jitter, tools for analyzing VoIP like connection tracking and packet sniffing, an example of making a SIP call, analyzing SIP call flows, marking packets for QoS, designing redundant VoIP networks, and using active monitoring and simulated packets. The presenter is an expert in MikroTik and VoIP with experience consulting and training.

Uploaded by

Naz Lun
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
331 views15 pages

VoIP in RouterOS PDF

This document provides an overview of mastering VoIP in RouterOS. It discusses codecs like G.711 and G.729 used for VoIP, protocol requirements around packet loss, latency and jitter, tools for analyzing VoIP like connection tracking and packet sniffing, an example of making a SIP call, analyzing SIP call flows, marking packets for QoS, designing redundant VoIP networks, and using active monitoring and simulated packets. The presenter is an expert in MikroTik and VoIP with experience consulting and training.

Uploaded by

Naz Lun
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
You are on page 1/ 15

Mastering VoIP in RouterOS

MUM Europe, Zagreb 2013.03.14


Presenter – Andis Arins 2

 MikroTik Consultant at www.router.lv

 MikroTik Certified trainer for all courses

 Member of the board in Latvian Internet Association

andis[at]router.lv
www.linkedin.com/in/andisarins
What is VoIP 3

1) analog -> digital


2) transmit over network
3) digital -> analog

Will It Work ?
Codec 4

G.711 64Kbps (precise speech transmission, low


CPU requirements) 87.2Kbps, 50pps, 160b payload

G.729 8Kbps (good speech quality, CPU costly,


requires license) 31.2Kbps, 50pps, 20b payload

GSM 13Kbps ( acceptable speech quality, available


in many hardware/soft platforms) ~36Kbps, 50pps
VoIP Protocol requirements 5

1) Packetloss (evenly < 0-5%)

2) Latency (< 120-150ms one way)

3) Jitter (buffer < 1-3ms)


Where to look for VoIP? 6

1) Connection tracking (conn-type SIP, Q931);


2) Torch tool
3) Firewall
4) Packet sniffer
Phone a friend 7

VoIP protocol: SIP, transport protocol UDP

R1 MUM R2 LATVIA
SIP

ssh

www.router.lv/va VOICE ANALYTICS


SIP call analysis 8

A - INVITE sip:[email protected] SIP/2.0


B – SIP/2.0 100 Trying
B - SIP/2.0 183 Session Progress
B – SIP/2.0 200 OK
A - ACK sip:[email protected] SIP/2.0

… RTP DATA …

B – BYE sip:[email protected];ob SIP/2.0


A - SIP/2.0 200 OK
Packet Flow and VoIP packets 9

Mangle Queue Tree Interface


200
byte 0.00004s – trip in RB750G

Queue Tree
Mangle
x

5M
LIMIT
1953 x 64 byte packets QUEUE SIZE

1M
83 x 1500 byte packets

3M
LIMIT
625 x 200 byte packets QUEUE SIZE
Mark a packet 10

1) src/dst interface
2) src/dst ip addres/port
3) connection-type
4) DSCP
5) layer7
QoS can travel with packet 11
QoS in RouterOS 12

tip - use only 80% of promised


VoIP design 13

ISP1 ISP2 ISP1 ISP2

LAN

1) Redundant internet (static/BGP)


2) Redundant router (VRRP)
3) Tunneled connection (SSTP, IPSEC)
VoIP design 13

Full redundancy, several links


Active monitoring with simulated packets
Thank You!

You might also like