1. VoIP / Internet Telephony
Peter Gradwell FBCS
26 Cheltenham St, Bath, BA2 3EX

2. Gradwell dot com Ltd Founded 1998 whilst an Undergrad
Now...30 Staff in Bath
3rd largest UK VoIP Provider for SMEs
30,000 lines on a VoIP platform gives us some interesting lessons to explore

3. Where does VoIP fit in?

4. Messaging + Talk

5. Different Types of VoIP

6. Voice Protocols
All systems use some kind of application protocol to communicate
SIP
H323
IAX (Inter Asterisk eXchange)‏
Skinny (Cisco Protocol)‏
Audio is transported as RTP

7. TCP/UDP/Ports Traffic is either TCP or UDP
MTU on Ethernet typically bytes
100 bytes of headers, bytes of data
Lots of packet overhead
Socket = IP Address + Port
MTU = Max Transfer Unit/A Packet

8. Good book: If you're interested in more details
You can read most of it on amazon

9. H.323 Point to Point Protocol H.323 is actually a family of protocols
H.323 is used a lot on trunk routes, as it's signalling reflects ISDN, but is becoming obsolete
Modem over IP
Conferencing
Fax

10. Inter-Asterisk eXchange
Asterisk is a major OpenSource phone system
IAX was invented by Mark Spencer/Digium
Single UDP stream – port 4569
Handles the Media inband on the same socket
Can support multiple RTP streams
Single signalling link
Far more bandwidth efficient
Also much friendlier for firewalls
Pronounced “eeks”

11. Skinny / SCCP
Skinny is a lightweight protocol which allows for efficient communication with Cisco CallManager
Not many other people use it, but there is a big Cisco install base
Signalling via TCP, Media via RTP/UDP

12. SIP – Session Initiation Protocol
SIP is the dominant VoIP protocol
It's a peer to peer protocol, unlike H.323

13. SIP Addressing SIP is very similar to Email and HTTP
URI Address format:

14. Types of SIP Message
INVITE—An INVITE method indicates that the recipient user or service is invited to participate in a session.
ACK - An ACK request confirms that the UAC has received the final response to an INVITE request.
ACK is used only with INVITE requests. ACK is sent end to end for a 200 OK response.
OPTIONS—A UA uses the OPTIONS request to query a UAS about its capabilities
BYE—A UA uses BYE to request the termination of a previously established session.
CANCEL—The CANCEL request enables UACs and network servers to cancel an in-progress request, such as INVITE. This does not affect completed requests in which the UAS had already sent final responses.
REGISTER—A client uses a REGISTER request to register its current location information corresponding to the AOR of the user with SIP servers.

15. SIP Registration
Phone registers periodically to tell the SIP server where it is

16. SIP Call Flow

17. A Complete Call
Often we'll draw SIP call path diagrams out like this for debugging

18. Debug using wireshark

19. You can also do redirects

20. SIP – Just for signalling
Register – Hello
Invite – Ringing
ACK – Please hold
OK – Connecting you caller
BYE – Nice to talk to you
But no sound!

21. Codecs
“Nyquist theorem states that if you sample an analog signal at twice the rate of the highest frequency of interest, you can accurately reconstruct that signal back into its analog form”
Audio is sampled at 8k or 16k times per second (Hertz) – Pulse Code Modulation
Codecs trade off quality, bandwidth and CPU to produce an RTP data packet
Narrow Band (8k) or Wide Band (16k)‏
We waste loads of Bandwidth on packet overhead
 Codec BR        NEB
 G.711      64 Kbps   87.2 Kbps
 G.729       8 Kbps   31.2 Kbps
 G.723.1   6.4 Kbps   21.9 Kbps
 G.723.1   5.3 Kbps   20.8 Kbps
 G.726      32 Kbps   55.2 Kbps
 G.726      24 Kbps   47.2 Kbps
 G.728      16 Kbps   31.5 Kbps
 iLBC       15 Kbps   27.7 Kbps
BR = Bit rate
NEB = Nominal Ethernet
Bandwidth (one direction)‏
Skype uses iLBC and ISAC

22. RTP RTP is used to send packets and re- assemble them What if?
Time stamp
Sequence
Payload type
What if?
They arrive out of order - Jitter
Or not at all? - Packet Loss

23. Jitter “Jitter is the variation of packet interarrival time”
Jitter is a major problem
The internet is asymmetrical
Manage the problem with a Jitter buffer ~ 20ms

24. Packet Loss Typically, each packet is 20ms of voice
If we loose one packet, it's not a major problem
Cisco implementation replays previous packet

25. Quality of Service QoS is needed if a network link is congested
Routers need to enforce traffic prioritisation across their links
How do they decide?
Trust the user? - DiffServ, Expedited Forwarding and Low Delay bits in IP header
Trust the network – all traffic on port X or from IP address Y is special

26. QoS Example

27. Phone numbers
Typically BT will route/transit the calls inbound from the world, to smaller Telcos
Could buy the service from C&W/Kingston
Call comes down an ISDN line (SS7 signalling) into a Server and we convert it to SIP
BT network is statically routed
Customer
Intl
Country
Area
Telco

28. ENUM Phone numbers as domain names
My VoIP server can now decide how to route the call
becomes
e164.arpa.
When I look this up I find
TEL: NNNNN
SIP:

29. ENUM Conceptually
Public ENUM is about publishing how you want to receive your phone calls
Carrier ENUM is telcos getting a dynamic routing table

30. SRV Records Can you dial sip:peter@gradwell.com
Yes – just do a lookup for the ENUM Records on gradwell.com and make a call to “peter”.
Peter =

31. Lets make a test call
login:
password: Jup5jojo
sip proxy: newsip.gradwell.net
Outbound proxy: natproxy.gradwell.net:5082
Display name: “test”
Then call your mobile – cli should be
Now dial inbound, everyones phone should ring!
This demo uses an ActiveX control, so only works on windows

32. Conclusions SIP – Signalling, RTP – Audio
But SIP isn't the only game in town
RTP is dependent on Network Quality
Routing information can be stored in ENUM

33. Useful Resources www.trixbox.org www.voip-info.org
Great VoIP wiki
Good Enum Primer
Cisco Book