snom voip primer christian stredicke june 2003. v1.0 2 2. h.323 3. sip 4. snom technology suite 5....

snom VoIP PrimerChristian Stredicke

June 2003

2

V1.0

2. H.323

3. SIP

4. snom technology suite

5. Installation of snom 4S

6. Conclusion

7. Test

1. Introduction to VoIP

3

V1.0

The Internet evolved from a research network into a real-time communications network

UDP

ftp

TCPDHCP

POP3

HTTP

SMTP (Email)

TLS

SIP

SNMP

IPv6 WAP

IM

Presence

PBX

RSVPDiffSrv

PPP

PPPoEPPTP

NATIPv4 DNS

RIP

AAAA

SRV

NAPTRENUM

SOAP

4

V1.0

The Internet Protocol (IP) is based on sending packets

EthernetHeader IP UDP RTP G.711 Ethernet

Checksum

Example anatomy of an audio packet

•Preamble•6 Byte Src•6 Byte Dst•2 Byte Type

•Version•Length•DiffSrv•ID•Flags•Fragment•Time to live•Protocol•Checksum•Source•Destination•20 Bytes

•Source Port•Dest Port•Length•Checksum•8 bytes

•Version•Padding•Extension•Source IDs•Marker•Payload Type•Sequence Number•Timestamp•Sync ID

• RTP = “Real-time transport protocol”• 54 Bytes header for every audio packet = 21.6 kBit @ 20 ms!• Packets can easily be viewed with Ethereal (www.ethereal.com)

5

V1.0

Sending voice packets is like sending freight with forwarders*

Sender in Germany

Receiver in India

LH901

LH710

Local Forwarder

Local Forwarder

Packet are sent with

Priority tag

May take different routes

May get lost during

transportation

Arrive with jitter, out of

order

* Just faster!

6

V1.0

Computer networks are designed for high bandwidth, but no delivery guarantee

• ISDN BRI = 128 kBit/s

• Ethernet 10 BT = 78 BRI

• Ethernet 100 BT = 781 BRI– At 39 € for a PC card!

• Ethernet 1000 BT = 7812 BRI

• WLAN 802.11– 11 MBit/s = 85 BRI– 54 MBit/s = 421 BRI

• 802.16

7

V1.0

RTP sometimes has to pass bottlenecks, especially on the network “edges”

Phone

DSL LineBackbone

TCP (http) may block B‘s RTP

traffic

Router

Customer can control outgoing

traffic

Router

TCPOther

RTP

RTP

Bandw

idt

h

Distance

Other

8

V1.0

Amongst the attempts to get telephony working on a IP-based network SIP seems to be the best

H.323

H.450

SIP (RFC2543)

SIP (RFC3261 ff.)

H.323v2

H.323v4

(Avaya etc.)

MGCP

MEGACO

(Skinny)ISDN over IP

9

V1.0

H.323 and SIP fight for customer acceptance

SIPH.323

• „Official“ standard

• Some devices in the market already („legacy devices“)

• Most devices do not support supplementary services (H.450)

• Similar to HTTP

• Less requirement on program space than H.323

• Suitable for large operators

• Cisco and Microsoft support SIP

• Darling of the Internet society

• Almost everybody jumping on the train

10

V1.0

Microsoft XP is going for the VoIP industry

• Getting Microsoft Windows XP means getting a SIP client

– 70 ms latency end to end, kernel support– Acoustic echo cancellation („hands free speaking“)– Presence list– Stability is ok– Microsoft-like nice user interface

• Windows XP embedded and Windows Pocket PC– Includes SIP stack as well as other features– Many gateways and stand alone equipment will this stuff

11

V1.0

2. H.323

3. SIP



6. Conclusion

7. Test


12

V1.0

Basic elements of an H.323 system

• Terminals– Telephones– Videophones– IVR devices– Voicemail systems– Softphones

• Gateways

• Gatekeepers

• Multipoint Control Units (MCUs)

13

V1.0

Gateways

• Gateways interface an H.323 network to other networks (ISDN,PSTN etc).

• Composed of Media Gateway Controller (MGC) and Media Gateway (MG).

– MGC handles all the call signaling functions. – MG handles the media related functions.

14

V1.0

Gatekeepers

• Optional Component of an H.323 system.

• Responsible for Registration, Address translation, Admission control, Bandwidth control and Zone management functions (RAS).

• Logical component of an H.323 system but Can be co-located within a gateway.

• Can perform direct or routed call signaling.

15

V1.0

Media Control Units (MCUs)

• Responsible for managing multipoint conferences (two or more endpoint engaged in a conference).

• The MCU contains a Multipoint Controller (MC) that manages the call signaling and may optionally have Multipoint Processors (MPs) to handle media mixing, switching, or other media processing.

16

V1.0

A typical H.323 network

17

V1.0

A simple H.323 call

18

V1.0

The big problem: H.323 is not really interoperable

Eff

ort

Features

ROSE

openh323

ASN.1

H.323

SIP

Proprietary

Private Small Business Professional• Most vendors are stuck in interoperability problems

• ASN.1 reduces productivity

• Most „implementers“ used openh323

• The star is falling

• Software concept works for stand alone PC, but can not be used on embedded systems

19

V1.0

2. H.323

3. SIP



6. Conclusion

7. Test


20

V1.0

3. SIP

3.1 User Point of View

3.2 Architecture

3.3 Registration

3.4 Basic Call

3.5 Call Control

3.6 PSTN interoperability

21

V1.0

The URL is the unique description of a telephony contact

• New URL types:– „Christian Stredicke“ <sip:[email protected]>– „Christian Stredicke“ <tel:[email protected]>

• Entering URL in hard phones is not so much fun– Some tricks to ease the pain– Automatic appending of domain– Address book, redial– Using TAPI for dialing from PC– Clever number setup („cs“, „np“, „tb“, ...)

• URL must always start with the “sip” scheme– May contain a large range of characters like +, -, %, .– Display names are encoded using UTF-8 (Unicode)

22

V1.0

SIP proxies search for the right person

• Registration includes information about probability of presence

– Moving the mouse changes the probability to 1.0– Leaving the mouse untouched lowers continuously the

probability– Mailbox has low probability (but above 0.0)

• Sequential forking:– Ringing one potential endpoint after another

• Parallel forking– Calling several parties at the same time– The first one picking up the call gets the call

• Phones can register for several accounts with different probabilities

23

V1.0

3. SIP


3.2 Architecture

3.3 Registration

3.4 Basic Call

3.5 Call Control


24

V1.0

The SIP architecture looks like a mixture of HTTP and Email

„Intranet“

„VPN“

Registrar 1

Registrar 2

Proxy 1

„Internet“

Proxy 2

Gateway 1

Gateway 2

UA

UA UA UAUA

UA

25

V1.0

The SIP proxy can be compared to a HTTP proxy: Taking complexity away from the client

• Finding a destination– Comparable to HTTP proxy– But real-time

• Requests may be redirected to other proxies

• Stateless proxies– Just forwarding packets

• Stateful proxies– Taking care of a call

• Other functions– Billing– Maintenance

• “Outbound proxy” handles all traffic of a dumb phone

26

V1.0

A stateless proxy just forwards packets to the right destination

Proxy

(1) INVITE(2) INVITE

(3) 100 Trying

(5) 180 Ringing

(4) 100 Trying

(6) 180 Ringing

(7) 200 Ok(8) 200 Ok

(9) ACK(10) ACK

27

V1.0

A stateful proxy may fork a call to multiple destinations

Proxy

(1) INVITE

(3a) INVITE(2) 100 Trying

(4b) 180 Ringing

(4a) 100 Trying

(5) 180 Ringing

(7) 200 Ok(8) 200 Ok

(10) ACK

(9) ACK

(3b) INVITE

(6a) 100 Trying

(6b) 180 Ringing

28

V1.0

Registrars remember where users can be found

• Close relationship with proxies– Usually registrar and proxy are the same program

• Registrars keep a list of registrations– One user can be found in different locations– Comparable to gatekeeper registration

29

V1.0

User agents (UA) are the endpoints of a SIP call

• Usually a hard or soft phone

• Gateways are user agents as well

• UA client (UAC) initiates something

• UA server (UAS) answers something

• UAS and UAC may change during a call– From and To-Fields are exchanges in this case

30

V1.0

SIP URLs provide a powerful way to describe a contact and its parameters

• Like HTTP URL

• Examples:– sip:[email protected]: Simple URL– <sip:[email protected]:5061>: Adding port number– “Fred F. Feuerstein” <sip:[email protected]>: Real name– <sip:[email protected]>;q=1.0: Probability of presence– <sip:[email protected]:5061;transport=tls>: Transport

layer– <sip:[email protected]>;q=1.0;expires=360: Several

parameters– sip:[email protected]?subject=project%20x&priority=urgent

• Details in RFC 3261

31

V1.0

Messages are encoding using ASCII characters (bit hackers go home!)

• printf() instead of ASN.1-Compiler

• Line termination with \r\n (tribute to MS-DOS)

• Double line termination means end of message body

• Attachments may follow of Content-Length indicates so

• Line continuation if new line starts with space

• See Email and HTTP.

Example:

REGISTER sip:[email protected] SIP/2.0From: „Christian Stredicke“ <sip:[email protected]> ;q=1.0To: „Christian Stredicke“ <sip:[email protected]>;q=1.0Cseq: 1 REGISTERContent-Language: enUser-Agent: snom V1.8bContent-Type: image/jpegContent-Length: 376

Jg45tzw49tz59p q3tz erihgisrgz dfkjh skdfgh hfg sdlfjgh sdfgjh sldfghsfghsdfhg lsfhg lskfgh sfkgh sfkgh slkfdg

32

V1.0

Principle of communication between entities

• Requests– Initiate something– Have something like „REQUEST sip:address SIP/2.0“ in the

first line of the message

• Response– Responds to a request– Look like „SIP/2.0 486 Busy Here“ in the first line– Codes < 200 are intermediate responses– Codes >= 200 and < 300 are success responses– Codes >= 300 are failure responses

• Acknowledge– Stands a little bit out in the communication– Terminates an INVITE transaction

33

V1.0

„Via“ headers trace the path of a request and allow a respond to travel that path back

• Every instance tags the path of a request taken so far by adding a via field

• Replies must be send back popping Via fields off the message

• Forking proxies add branch parameters to the via fields to tag different branches

Via: host1

Via: host2Via: host1

Via: host3;branch=123.1Via: host2Via: host1

Via: host3;branch=123.2Via: host2Via: host1

34

V1.0

„Record-Route“ fields are used to ensure that some proxies are part of subsequent requests

• SIP does not mandate that requests between two UA take the same path

– Actually, SIP encourages taking a direct path even for ACK

– That causes problems with stateful proxies and billing systems

• Record-Route ensures that proxies that add themselves to the route are in the message chain in future requests

1

2

3

4

Route: h2, h4

Route: h4

5

Contact: h1

Record-Route: h2Contact: h1

Record-Route: h2Contact: h1

Record-Route: h2, h4Contact: h1

1st request Subsequent requests

35

V1.0

Every message has a sequence number

• CSeq field– CSeq: 1 REGISTER– CSeq: 1 INVITE

• Sequence numbers are kept within:– A call– A registration– A notify, subscribe, ...

• UAC and UAS keep their own sequence numbers for requests

• If a client receives a message with an CSeq number less or equal to the one received, it is takes an message repetition

– Relevant for UDP

36

V1.0

SIP can be transported over UDP, TCP and TLS

• UDP– Messages may get lost– Mandatory– Repetition of messages:

– Requests until a response arrives (same CSeq)– Responses if a request repetition arrives– ACK if a response arrives twice

• TCP– Like HTTP– Content-Length becomes very important

• TLS– Is a SSL (Secure Socket Layer) version– Keeps SIP traffic secure– Like TCP– Different port to avoid misunderstandings (e.g. 5061)

37

V1.0

To save some space, short names of header fields are available

• Saving space makes up 0.000143 % of an average call

– In other words, does not make sense– May help to keep the UDP message size boundary

(but not really)

• However, feature is mandatory– From = f– To = t– Content-Length: l– Call-ID: I

• See draft

38

V1.0

3. SIP


3.2 Architecture

3.3 Registration

3.4 Basic Call

3.5 Call Control


39

V1.0

Registration is a simple mechanism to give the UA an identity

• UA sends REGISTER request to registrar (or outbound proxy)

• Registrar answer with response– 200 Ok: Registration is ok– 404 Not Found: User may not register here– 403 Not Found: User is not known– 100 Trying: Intermediate response– 401 Authentication Required: UA has to provide a

password– Any Response!

40

V1.0

Example 1: Successful registration

• Phone wants to register as “1” at registrar “62.254.248.7”

• Address does not need to be a dots-and-number address

• Registrar returns all registrations for this number

• Expiry time indicates how long the registration will last and is determined by the registrar

• Example taken from Cisco proxy registration

REGISTER sip:62.254.248.7 SIP/2.0From: <sip:[email protected]>To: <sip:[email protected]>Call-ID: [email protected]: Thu, 16 Aug 2001 17:29:22 GMTCSeq: 1 REGISTERContact: <sip:[email protected]:5060;transport=udp>Via: SIP/2.0/UDP 62.254.248.56:5060Expires: 3600Content-Length: 0

SIP/2.0 200 OKVia: SIP/2.0/UDP 62.254.248.56:5060Call-ID: [email protected]: <sip:[email protected]>To: <sip:[email protected]>CSeq: 1 REGISTERContact: <sip:[email protected]:5060;transport=udp>;expires="Thu, 16 Aug 2001 15:31:14 GMT"Contact: <sip:[email protected];method=INVITE>;description="OPEN";expires="Thu, 16 Aug 2001 21:57:01 GMT"Expires: 3600Content-Length: 0

41

V1.0

Example 2: Failed registration

• Phone wants to register as “1” at registrar “a1.sipbakeoff.org”

• Registrar responds with intermediate message 100 Trying

• 400 Code indicates that something went wrong

• 403 means that there was no account set up at the registrar

REGISTER sip:a1.sipbakeoff.org SIP/2.0From: <sip:[email protected]>To: <sip:[email protected]>Call-ID: [email protected]: Thu, 16 Aug 2001 17:29:01 GMTCSeq: 1 REGISTERContact: <sip:[email protected]:5060;transport=udp>Via: SIP/2.0/UDP 62.254.248.56:5060Expires: 3600Content-Length: 0

SIP/2.0 100 TryingVia: SIP/2.0/UDP 62.254.248.56:5060From: sip:[email protected]: sip:[email protected]: [email protected]: 1 REGISTERContent-Length: 0

SIP/2.0 403 - ForbiddenVia: SIP/2.0/UDP 62.254.248.56:5060From: sip:[email protected]: sip:[email protected]: [email protected]: 1 REGISTERContent-Length: 0

42

V1.0

Example 3: Authentication challenges the UAC with a 401/407 response

• Registrar denies registration with 401 Authentication Required

• 401 contains challenge that must be answered by UAC

• UAC tries again with Authentication-Field

• Seconds register is new request, therefore CSeq is increased

• In general, every request can be challenged

– Multiple times, multiple lines

– Digest mandatory, Basic discouraged

REGISTER sip:a1.sipbakeoff.org SIP/2.0From: <sip:[email protected]>...Cseq: 1 REGISTERContent-Length: 0

SIP/2.0 401 Authentication RequiredWWW-Authenticate: Digest realm="a.com",domain="sip:[email protected]:7072;maddr=62.254.248.14",nonce="aWGQ03+9PIZNYfTthPUSlA==",algorithm=MD5Cseq: 1 REGISTERContent-Length: 0

REGISTER sip:a1.sipbakeoff.org SIP/2.0From: <sip:[email protected]>...Authorization: Digest username="01",realm="a.com",nonce="9wTTyVG1WlOwkdpHBEmioQ==",response="d39db66212c999fbb0f1c767364479eb",uri="sip:62.254.248.14:5070"Cseq: 2 REGISTERContent-Length: 0

SIP/2.0 200 Ok...Cseq: 2 REGISTERContent-Length: 0

43

V1.0

Which information does a client has to set up for port forwarding in NAT equipment?

• Router needs information where to send packets in private network

– Map port to private address and port

– By default packets will be rejected or sent to DMZ

• Router needs hint for security checking

– Accept packets from any destination

– Accept packets only from associated host

– Accept packets only from associated host and port

12

3.1

23

.12

3.1

23

19

2.1

68

.0.1

Router

Client

Client

44

V1.0

STUN uses the digging hole trick to set up port associations

• Initialization procedure checks environment– Goal: Check if STUN is needed– Type of NAT does actually not really matter because user

is not interested in failure reason

• SIP port kept alive by sending packets every 15-60 s

• RTP ports are allocated dynamically when starting a call

– Otherwise keep-alive traffic would be double– RTCP port can not be allocated because next port

allocation is unlikely– Long ringing and putting caller on hold is problematic (no

port refresh during this time)

45

V1.0

How does port forwarding in UPnP work?

• Find the Internet access device– Broadcast messages (no user setup required)– Download the description of the UPnP device via http

• Retrieve the public IP address from the router

• Set up port mapping explicitly– http requests using XML (SOAP) attachments

• Other commands also available– UPnP is much more than setting up port forwarding on

routers

46

V1.0

Registering Procedure of the snom phones

1. Send a REGISTER message with the IP unchanged identity (coming from DHCP or static setup)– Includes a rport parameter in the top via– This parameter get completed if the proxy receives the packet

from a IP address that is different from the unchanged identity– The proxy sends the packet to the port indicated by rport (that’s

how the packet makes it back to the phone)– If the phone does not receive the changed rport back, the

registration is done (public IP address, ALG, inside private network registration)

2. If there is a rport parameter returned, the phone will try to do UPnP or STUN– Default policy is to try UPnP first, then STUN– Can be changed in the settings

3. If the public IP address has been determined, the phone will register with this– Then there is no more checking NAT method

47

V1.0

3. SIP


3.2 Architecture

3.3 Registration

3.4 Basic Call

3.5 Call Control


48

V1.0

INVITE is used to establish a call

(1) INVITE

(2) 100 Trying

(3) 180 Ringing

(4) 200 Ok

(5) ACK

• UAC sends INVITE

• UAS may respond with 100 Trying

– If expected answer takes a while

• UAS may respond with a 180 Ringing

– Indicates that the UAS is ringing the user

• UAS sends 200 Ok if the connection is set up

• UAC sends ACK to confirm that session has been set up

49

V1.0

Example 4: Successful INVITE: Sending out the INVITE

• First line contains destination

– May contain tags

• UAC tags from-field that contains the origin

– however this address is not used for reaching the originator

• Call-ID should be unique in universe

• Contact field indicates where requests may be sent

• SDP attachment describes where audio may be sent

INVITE sip:[email protected] SIP/2.0From: <sip:[email protected]:7071>;tag=hsuiykhtpfTo: sip:[email protected]: SIP/2.0/UDP 62.254.248.56:5060Call-ID: [email protected]: 1 INVITEContact: <sip:[email protected]:5060;transport=udp>User-Agent: snomAccept-Language: enAccept: application/sdpSupported: sip-cc, sip-cc-01, timerSession-Expires: 120Content-Type: application/sdpContent-Length: 236

v=0o=root 26500 26500 IN IP4 62.254.248.56s=Call from 62.254.248.56c=IN IP4 62.254.248.56t=0 0m=audio 5000 RTP/AVP 0 3 8 101a=rtpmap:0 pcmu/8000a=rtpmap:3 gsm/8000a=rtpmap:8 pcma/8000a=rtpmap:101 telephone-event/8000

50

V1.0

Example 4: Successful INVITE: Receiving provisional responses

• 100 Trying– Probably came from proxy– Stops repeating UDP

messages

• To field still untagged in this case

– It may be tagged and then we have a call leg

– If tagged the call splits into main call (untagged) and legs

• 180 Ringing– UA may play ring back now

• Provisional responses may be acknowledged by PRACK

– Capability to receive PRACK has to be indicated in Allow header

SIP/2.0 100 TryingTo: sip:[email protected]: <sip:[email protected]:7071>;tag=7q30roe94yVia: SIP/2.0/UDP 62.254.248.56:5060Call-ID: [email protected]: 1 INVITEContent-Length: 0

SIP/2.0 180 RingingTo: sip:[email protected]: <sip:[email protected]:7071>;tag=7q30roe94yVia: SIP/2.0/UDP 62.254.248.56:5060Call-ID: [email protected]: 1 INVITEContent-Length: 0

51

V1.0

Example 4: Successful INVITE: Receiving 200 Ok

• To field now contains a tag– Call now being split up into

legs

• Contact field indicates where further requests may be sent

• SDP attachment contains media data

SIP/2.0 200 OK To: sip:[email protected];tag=150148666From: <sip:[email protected]:7071>;tag=7q30roe94yVia: SIP/2.0/UDP 62.254.248.56:5060Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]: application/sdpContent-Length: 153

v=0o=deepa 134278 968037671 IN IP4 62.254.248.105s=IPNessc=IN IP4 62.254.248.105t=0 0m=audio 5000 RTP/AVP 0a=rtpmap:0 PCMU/8000a=ptime:30

52

V1.0

Example 4: Successful INVITE: Sending ACK

• CSeq number remains unchanged

– ACK is not a new request

• In this case the ACK is directly sent to the address given in the Contact field

– Obviously, no proxy requested Record-Routing

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 62.254.248.56:5060From: <sip:[email protected]:7071>;tag=7q30roe94yTo: sip:[email protected];tag=150148666Call-ID: [email protected]: 1 ACKUser-Agent: snomContent-Length: 0

53

V1.0

Example 4: Successful INVITE: Hanging up

• BYE request directly sent to Contact

– Alternative would be using outbound proxy

• To and From-fields just copied

– If other party hangs up, To and From has to be exchanged

• CSeq in increased– But only on the UAC side!

• All other requests except INVITE and not acknowledged by ACK

BYE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 62.254.248.56:5060From: <sip:[email protected]:7071>;tag=7q30roe94yTo: sip:[email protected];tag=150148666Call-ID: [email protected]: 2 BYEContent-Language: enContent-Length: 0

SIP/2.0 200 OK To: sip:[email protected];tag=150148666From: <sip:[email protected]:7071>;tag=7q30roe94yVia: SIP/2.0/UDP 62.254.248.56:5060Call-ID: [email protected]: 2 BYEContent-Length: 0

54

V1.0

Example 5: Canceling a call (1)

• INVITE, 100 Trying and 180 Ringing like before

• User decides to cancel call, UAC sends out CANCEL

– CSeq number remains unchanged but method becomes CANCEL

– To field does not contain tag

• UAC receives 200 Ok on CANCEL

– This is only on the CANCEL!

CANCEL sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 62.254.248.56:5060From: <sip:[email protected]:7071>;tag=qnbibiw4zjTo: sip:[email protected]: [email protected]: 1 CANCELContent-Language: enContent-Length: 0

SIP/2.0 200 OkTo: sip:[email protected]: <sip:[email protected]:7071>;tag=qnbibiw4zjVia: SIP/2.0/UDP 62.254.248.56:5060Call-ID: [email protected]: 1 CANCELContent-Length: 0

55

V1.0

Example 5: Canceling a call (2)

• UAC receives 487 on INVITE– Therefore, the INVITE

failed

• If UAC receives 200 Ok on INVITE, the call is established

• Anyway, the UAC has to acknowledge the final result with an ACK

SIP/2.0 487 Request TerminatedTo: sip:[email protected]: <sip:[email protected]:7071>;tag=qnbibiw4zjVia: SIP/2.0/UDP 62.254.248.56:5060Call-ID: [email protected]: 1 INVITEContent-Length: 0

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 62.254.248.56:5060From: <sip:[email protected]:7071>;tag=qnbibiw4zjTo: sip:[email protected]: [email protected]: 1 ACKUser-Agent: snomContent-Length: 0

56

V1.0

3. SIP


3.2 Architecture

3.3 Registration

3.4 Basic Call

3.5 Call Control


57

V1.0

Hold is just a modification of the SDP record that has been sent to the other party

• To put the other party on hold, either party may send a re-INVITE

– The IP address 0.0.0.0 indicates the other party that it should stop sending media

• Re-INVITE must be replied to with a 200 Ok

– 100 Trying not necessary– 180 Ringing not allowed

here

• UAC answers with ACK

• 200 Ok may contain SDP record as well

– In order to allow music on hold, it should not do that

INVITE sip:[email protected]:5060;transport=udp SIP/2.0Via:SIP/2.0/UDP 62.254.248.6:5060;branch=bw.62.254.248.6From:<sip:[email protected]>;tag=247792666-997982268876To:<sip:[email protected]:5060>;tag=oe94ywxqnbCall-ID:[email protected]:774927797 INVITEContact:<sip:62.254.248.6:5060>Allow:ACK,BYE,CANCEL,INFO,INVITE,PRACK,REFERSupported:100rel,timerContent-Length:210Content-Type:application/sdp

v=0o=CiscoSystemsSIP-IPPhone-UserAgent 6734 5093 IN IP4 62.254.249.87s=SIP Callc=IN IP4 0.0.0.0t=0 0m=audio 16392 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0

58

V1.0

Call forwarding is implemented using 301 and 302 responses

• 301 and 302 are treated like error codes

– Transaction is acknowledged and over

– That implies that the new transaction may have a new CSeq

• A new INVITE is sent to the given contact

• 301 and 302 are business of proxies

– In the context of parallel and sequential forking

– Multiple contacts cause trouble on most UA

INVITE sip:[email protected] SIP/2.0From: <sip:[email protected]:5060>;tag=nxawl7blpmTo: sip:[email protected]: SIP/2.0/UDP 62.254.248.57:5060Call-ID: [email protected]: 1 INVITEContact: <sip:[email protected]:5060>;transport=UDP...

SIP/2.0 302 Moved temporarilyVia: SIP/2.0/UDP 62.254.248.57:5060From: <sip:[email protected]:5060>;tag=nxawl7blpmTo: sip:[email protected]: 1 INVITECall-ID: [email protected]: sip:[email protected]

ACK sip:[email protected] SIP/2.0...

INVITE sip:[email protected] SIP/2.0From: <sip:[email protected]:5060>;tag=nxawl7blpmTo: sip:[email protected]: SIP/2.0/UDP 62.254.248.57:5060Call-ID: [email protected]: 1 INVITE...

59

V1.0

Transferring a call using REFER (1)

(1) INVITE (hold)

(6) INVITE

(9) NOTIFY

(7) 200 Ok

(2) 200 Ok

(3) ACK

(4) REFER

(5) 202 Accepted

(8) ACK

(10) 200 Ok

(11) BYE

(12) 200 Ok

60

V1.0

Transferring a call using REFER (2)

• First put the call on hold (1-3)– Otherwise the other phone could get into trouble having

only one media stream (would have to put the call on hold as well)

• Request a transfer with REFER– 202 Response stops retransmission of the request– Other party now tries to set up the requested call

• Notify originator about transfer result– REFER request is already over, so a new request must be

initiated be the other party– Attachment contains result

• If everything went ok, the original call can be closed– If not, the original call be be reactivated

61

V1.0

To know which state another UA is in, UA should subscribe to Dialog-State

• Similar to presence, but more phone specific– States for ringing and on hold

• XML-Based coding– State information– Call-ID, From, To

• Important for Call Pickup

• Important for LED control

62

V1.0

Music on hold, call park, call pick & friends

• Are implemented using the methods above

• Music on hold invites the media server to the other party

• Call park initiates transfer to park server– Sometimes this is also called “hold”

• Call pick is done by a INVITE with the Replaces header– Information known by dialog state-subscription

• No introduction of new messages for call control

63

V1.0

NOTIFY is a simple and powerful way to send information to a UAS

• NOTIFY carries “attachment”– Could be the result of a REFER– Could be presence status

• Type is indicated in Content-Type

• Uses the transport mechanism of SIP like all other messages

• Has From, To, Call-ID, CSeq, Content-Length like all other messages

• Is subject to the message repeating mechanism like all other messages

64

V1.0

Message waiting indication (MWI) is a good example for the usage of the NOTIFY mechanism

• NOTIFY goes through proxy as unknown message

– All registered users (including mailbox) receives NOTIFY

– All users answer and effectively stop sender from repeating

• Whoever reads the message responds with a 200 Ok

NOTIFY sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 62.254.248.89:5060From: <sip:[email protected]>;tag=19140-10589-62.254.248.89To: <sip:[email protected]>CSeq: 21911 NOTIFYCall-ID: [email protected]: application/simple-message-summaryEvent: simple-message-summaryContent-Length: 48

Message-Waiting: yesVoicemail: 3/0Fax: 1/0

SIP/2.0 200 OKFrom: <sip:[email protected]>;tag=19140-10589-62.254.248.89To: <sip:[email protected]>Via: SIP/2.0/UDP 62.254.248.89:5060Call-ID: [email protected]: 21911 NOTIFYUser-Agent: snom V1.6cContent-Length: 0

65

V1.0

3. SIP


3.2 Architecture

3.3 Registration

3.4 Basic Call

3.5 Call Control


66

V1.0

Overlap dialing is implemented using the 484 response

(1) INVITE (CSeq = 1)

(6) INVITE

(7) 180 Ringing

(2) 484 Address Incomplete

(3) ACK

(12) ACK

ProxyGateway

(4) INVITE (CSeq = 2)

(5) 100 Trying

(11) ACK

(8) 180 Ringing

(9) 200 Ok(10) 200 Ok

„IAM“ Initial Address Message

„SAM“ Subsequent Address MessageTimeout

67

V1.0

Provisional acknowledge (PRACK)

• Problem: Provisional responses (1xx) are not reliable in UDP

– This important because temporary provisional media streams need acknowledge

• Solution: Acknowledge them with PRACK

• PRACK has its own sequence number (RSeq)

• See example

INVITE sip:[email protected] SIP/2.0Supported: 100rel...Content-Type: application/sdpContent-Length: 236

SIP/2.0 183 ProceedingRequire: 100relRSeq: 776655...CSeq: 1 INVITEContent-Length: 0

PRACK sip:[email protected] SIP/2.0RAck: 776655 1 INVITE...CSeq: 2 PRACK

SIP/2.0 200 OKVia: SIP/2.0/UDP saturn.bell-tel.com...CSeq: 2 PRACK

68

V1.0

Temporary provisional media streams (183)

(1) INVITE (sdp = …)

(2) 100 Trying

(3) 183 Session Progress

Gateway

(4) PRACK

(5) 200 Ok (PRACK)

(Ringback RTP)

(6) 200 Ok

(7) ACK

(2 way audio)

• Initial INVITE indicates where audio is expected

• Gateway sends 183 Session Progress and starts playing audio one way

• UAC confirms provisional with PRACK

• When other party picks up, 200 Ok is send and UAC starts sending audio as well

69

V1.0

2. H.323

3. SIP



6. Conclusion

7. Test


70

V1.0

The snom 4S consists of a real-time part (media server) and a call control part

Fast, but not too

intelligent

Intelligent, but

sometimes takes a

little timeProxy

MediaServer

71

V1.0

The proxy consists of several components that are necessary for SIP

Registrar Proxy Location Server

Scripting

Web Interface

Billing

DNSXML

Logging

72

V1.0

The proxy runs different domains completely separately

• Like renting a hotel room– Different customers with different domains (snom.com, bnc.co.jp)– Test and productive system

• Customers may access the proxy with their private password

• Data is kept separate– Customer A cannot see what customer B is doing

• Enables central IP services

73

V1.0

Stateful information is permanently stored in the file system

• Stateful Information– Registration Information– Messages– But not ongoing calls

• XML– Human readable/editable format– Can be used together with revision control systems

(CVS)

• Allows large scalability– 1 Million users fit easily on modern hard disk– Performance then is the problem

74

V1.0

The scripting makes customization and programming of the proxy simple and safe

• Copy success story of PHP (Apache)– Easy to understand– Extremely flexible– Cannot crash proxy so easily

• No CPL (Call Processing Language)!– Too complicated– Not powerful enough

• Web Interface sets variables that are interpreted by default script

– Quick setup for most typical cases– No programming necessary

75

V1.0

NAT support has great important in practical usage

• Reject private address registration– Give a hint using the rport parameter– Frustrating history of users trying to register their phone

behind NAT

• STUN server included– Only necessary functionality

• Support of path registrations– Clients register with a list of addresses instead on only one

76

V1.0

Instant messaging is not the job of the proxy; however the proxy does its best to support it

• Normal request handling rules– Implements full IM functionality

• Store & Forward– Store messages on file system until user registered again– Simple & effective

• Welcome Message – When users sign in first time they get a greeting message

77

V1.0

Presence can be done on the user agents; caution has to be taken about performance

• Number of presence updates can easily kill the proxy– 10000 users with 50 contacts each changing their state

every hour generate more than 100 transactions per second!

– Questions the presence promise

• We wait until a feasible solution for this problem has been identified

• When user agents support presence (like Microsoft Messenger) proxy can handle it

78

V1.0

Billing: One of these tricky issues…

• Has been “forgotten” by SIP in the beginning

• Interfaces:– Plain file (named pipe, …)– XML billing notifications via SIP– RADIUS

• Problems– Bypassing proxy (solution for PSTN termination: gateway

setup)– Caller information– Hangup upon account expiration

79

V1.0

ENUM allows the lookup of telephone numbers to DNS names

• Simple mechanism– tel:+493039833401– 1.0.4.3.3.8.9.3.0.3.9.4.arpa.net– Service pointer for this DNS entry

• Today there are some problems– Overlap dialing does not work with ENUM– Security is hard to ensure (redirecting to expensive PSTN

numbers)

• snom 4S offers ENUM– Secure DNS missing (will be available at end of March)

80

V1.0

The user may access the proxy to see the call list, messages, redirection settings, passwords, etc.

• See missed calls even if not registered

• Independent from location where phone is

• Simple redirection– Simple setup for redirecting to cell phone– Voice mail integration with different account number

81

V1.0

The media server is based on a generic XML-programmable subsystem

• Simple description of the various media types– Variables– Substitutions for date, number of messages etc– Simple commands like store message

• Programming errors cannot crash the media server

• Performance is very good– XML is converted to internal table

• Users can set up their own media types– Adding new language– Modifying existing types– Creating new types

• Subsystem supports transcoding– Conference between different codec calls

• Using the snom phone SIP stack

82

V1.0

We offer the most typical media types in the standard installation

Mailbox Auto Attendant

Conference

Music, Error

Calling Card

Sample mailbox with different modes (personal greeting, standard greeting, absence, etc.)

Auto attendant initiates transfer of existing call or early media call (REFER or 3xx redirect).

Simple conference bridge with or without authentication of users. Can also be used for callback and click-to dial and B2BUA.

Music on hold server. Reading error explanations.

Simple Calling Card application with precise billing.

83

V1.0

Performance measurements have shown good performance for G.711 and caution for G.729A

0

0,1

0,2

0,3

0,4

0,5

0,6

0,7

0,8

0,9

1

2 G729

4 G729A

20 G711

30 G711

10 G711 Conf

750 MHz Pentium II

84

V1.0

Calling Cards…

• Main problem: What happens when card becomes empty?

– User continues talking– User is not warned– User may want to hear a gong for every coin he uses

• Solution: Calling Card in media server– Media server keeps track on used coins– No multiple call problem– Media server generates tones for billing information

85

V1.0

Scaling snom 4S

• SIP can inherently be scaled– If user agents support DNS SRV– Proxy supports database replication

• Performance is not the issue on the proxy– 1 Mio calls per day results in a approx. 50 transactions per

second (no problem)

• Media Server needs to

86

V1.0

We run our proxy and media server for more than a year now at snom.com

• Several domains– snom.info for snom 200 phones Plug & Play– snomag.de as free proxy for everyone who wants to try– snom.com for our telephony traffic– stredicke.de for sip:[email protected]

• Pubic media accounts– sip:[email protected] for sample conferencing– sip:[email protected] for test calls– sip:[email protected] for Internet music on hold– Plug & Play setup of snom.info mailboxes

• Exposed to the public Internet– We had a few DoS attacks– But no problems with the proxy stability

• Performance seems not to be an issue– Currently, approx. 1000 users are actively using the proxy

87

V1.0

The snom 105 is a small-size phone with a graphical display

• Optimized design for VoIP– Connectors on the back– 128 x 64 B/W display

• Cellular phone like functions– Function keys– Animated menus– Message display

• Asian character support

• Multiple registrations, multiple calls

88

V1.0

The snom 200 is a office phone with a two-line display

• Phone for business area– LED keys– Dedicated keys for special

functions– Large handset, good audio

performance– 2-line display

• More simple to produce

• Smart functions– Number guessing– Remote state indication

89

V1.0

The embedded web server make access to the proxy easy

• Complex setup can be done via the web browser– Line registrations– Diagnostics– Remote management– Address book management

• Dialing from the web browser– Helpful for SIP URL

• Web browser allows remote control– Also from other applications– E.g. TAPI (Outlook)

90

V1.0

“Plug and Play” support makes mass deployment easy

• Large installations must have a mechanism that automatically installs the phones

• No new mechanism should be used for SIP and VoIP

• snom uses http for transporting the configuration data– Existing infrastructure can be used (Apache, mysql)– Simple ASCII based format– Tunnels trough firewalls and NAT– Settings can be made read-only

• Phones can be put into operators realm– Redirect according to MAC address– Users may select operator realm from web interface

• Automatic software update

91

V1.0

2. H.323

3. SIP



6. Conclusion

7. Test


92

V1.0

Installation on Windows Platforms can be done easily if you follow some rules

• Just follow the dialog and the manual!

• Be sure to be administrator– Otherwise you might have problems adding the proxy as a

new service– In this case, the proxy does not start automatically

• You don’t need to reboot the machine– Manually starting from the service manager– But make sure that after a reboot the proxy is coming up

• Check the event log for extraordinary messages

93

V1.0

Windows stores some basic information in the registry and then reads the rest from the files

• Get the root directory from the registry– HKEY_LOCAL_MACHINE/SOFTWARE/snom/proxy is the root

path– reg_dir contains the root directory for the proxy– html_port_number and sip_port_number define the ports

• Other configuration information is stored in config.xml– Configuration of the “admin” account– Domains are stored in the domain directories– Don’t edit this file when the proxy is running (changes will

overwrite it)

94

V1.0

Updating the Windows service is not so easy, you have to follow some voodoo-tricks (warp II)

• Mark the service for manual starting– Keeps Windows from using this file after reboot

• Reboot

• Uninstall the old software– Is now safe because image is not being used

• Install the new software– Start it in the service manager

• Check the version number and the build– If they did not change you have a problem

95

V1.0

The install.sh script simplifes the Linux installation for SuSE and RedHat

• Follow the descriptions in the manual

• Make sure you are “root”

• The script sets up run level information and copies the executable into the /usr/sbin directory

– Be careful with name mixing (rcsnomproxy, snomproxy)– Old proxy was named sip-proxy, new has name snomproxy

96

V1.0

RedHat startup procedure

• /etc/snomproxy.conf contains the variables for startup information

– START_SNOMPROXY: yes or no– SNOMPROXY_OPTS contains the command line interface

arguments– This usually contains the html/sip port and the startup

directory– snomproxy.conf is overwritten during (re-)installation

• Use /etc/init.d/snomproxy stop/start for controlling the proxy

97

V1.0

Should the customer use Linux or Windows?

• If the customer has a preference, don’t try to convince him

– Proxy runs fine on both platforms– Performance is good on both systems

• Linux is easier to access with telnet/SSH– For operators that can be an advantage– SuSE is “European” style, RedHat “American”– We had some trouble with RedHat, now it’s a well

supported platform– Try to stay away from other distributions as we don’t

support them

• Windows can be managed by most administrators– For companies that can be an advantage

98

V1.0

Licensing keeps people from using illegal copies of our software

• Licensing is bound to the IP address– The current IP address of the proxy can be seen on the

Licensing Web page (on the top)

• The proxy must match the IP address to one of the hostnames

– All license names must be in the hostname list– Later we do DNS resolution on the hostnames as well

• If the proxy is not licensed it rejects all requests with “500 License Expired”

– But the rest of the proxy is operative

99

V1.0

You can go to the domain by clicking on the link or by logging in

Log in as:•Administrator•Domain Administrator•User

100

V1.0

Checking if the SIP proxy is up & running

• Check the product version– Version 2.xy (x = major, y = minor release)– Version 2.xyz (z = beta release indicator)

• Check if the service comes up after reboot

• Check if the product is licensed

101

V1.0

First you should make sure that the fundamental settings are ok

Choose Ports

Use this only if theproxy runs on several

IP addresses

Logging helps identifying problems

Set a password

Leave this emptyin the beginning

102

V1.0

Before you can start operating the proxy, you need to set up a domain

Click here to go to the domain

Enter a domain name here to create a new

domain

Edit or remove an

existing domain

103

V1.0

Take a look at the domain settings first and set the password for the domain administrator

Set your domainpassword here

Enter the linkto the mediaserver (see

below)

104

V1.0

Define the way the proxy handles registrations

Allow only knownusers to register

Define the expiration

Default probability for user agents

Avoid problems with devices behind NAT

105

V1.0

Set up your first user account

Name of the account Name for authentication (usually

the same as the account name)

Choose password that cannot be guessed easily

Other names for the account

Single registration avoids problems

when clients register with

changing addresses

106

V1.0

Check if your account has been set up

Name of the accountClick here to edit this

account

Number of current registrations on this

account

Remove the accountAccounts can be

enabled or disabled

107

V1.0

On the phone, set up the registration information as well (credentials first)

Define the expiration

Default probability for user agents

Define for which lines the line is valid

Typically the domain name

Username as defined on the proxy (not the

account!)

Password

108

V1.0

Enter the account information in the SIP/Line Settings menu on the phone web interface

Choose on of the accounts

Put your „real“ name in here

The account name of the

proxy (not the username)

Domain name

Leave as is Leave as is

Mailbox address (leave empty for

now)

Probability that user can be found

on this phone

109

V1.0

Check if the registration succeeded

Current local time

How the phone can be reached, maybe including the path to the

user agent

Phone type

„Real“ name

Domain

Remove this registration (account is not deleted)

Duration of registrationAccount name

110

V1.0

Looking at the SIP trace you also get an idea what's going on

Local time of reception

IP addressPacket behind

link

UDP or TCP

Rx = Received (normally)Tx = Send (normally)Rr = Repetition receivedTa = Send automatic answerTr = Send message repetitionTf = Send failed

„Good“ registration

„Good“ registration,

message repetition

111

V1.0

After making your first call, you should see the call in the call trace

Local time of callSource; clicking

here show details of the call

Destination

Duration

112

V1.0

A typical call flow includes forking the request to several destination

Initial Invite

Phones show that they are ringing

Forking to phone and

mailbox

Mailbox picks the calls up

Proxy cancels the call to the phone

Hang up

Acknowledgement of cancel

113

V1.0

A simple dial plan for a office with a single gateway

Mode is always forward

All calls with at least 4 digits go to the gateway

Redirect calls to the FAX to

another number

Incomplete incoming calls (at

least 5 digits) go to central number

114

V1.0

Before operating the media server, you should check the fundamental settings

Make sure the IP address is correct If you want to use

Email, enter the server information here

Making entering number more

simple

Make sure these ports are available

Avoid CPU overload

115

V1.0

The stack settings of the media server normally need no change

Only necessary if you are using NAT

Set during installation

Switching between transport layers

Automatic hang-up when clients don’t

reconnect

Message repetition (in ms)

116

V1.0

Also, the codec settings normally don’t need to be changed

U-law has best quality, but high bandwidth

Packet delay is important only for

conferencing

Media server automatically

detects DTMF, this setting is about indicating thisPayload type is

normally negotiated

Force a specific codec

117

V1.0

The currently known accounts on the media server

Name of the account Delete an account

Type of the account

Edit an account

118

V1.0

You can add accounts by selecting its type and account name

Name of the account Type of the account

119

V1.0

The music on hold account type just plays music; the user may select the type with DTMF

Authentication information

All accounts have a language setting

Location of the music files

All accounts have a password

Where the account should register

120

V1.0

The conferencing account mixes the audio streams of all participants


If password is present, the caller is

prompted for the password before he

can join the conference

121

V1.0

The conferencing account mixes the audio streams of all participants

Time until the mailbox picks up


If this matches the To-header, the

mailbox does not ask for the password

After recording a message, the

mailbox sends a MWI to this destination

Maximum recording length

122

V1.0

<?xml version ="1.0" standalone="yes"?><media_definition>  <option name="x-type">Mailbox2</option> <option name="x-conference">false</option> <option name="x-msg_send_mwi">true</option> <option name="x-msg_send_email_summary">true</option> … <option name="email_url"></option> <option name="email_name"></option> <option name="x-email_file"></option>

 <start condition="equal ${owner} true">welcome_main_menu</start> <start condition="equal ${x-mailbox_mode} mb_name">mailbox_enabled_name</start> <start>mailbox_enabled_stand</start> 

 <state name="mailbox_enabled_stand">  <audio type="file">${audio}/${lang}/mb_you_have_reached_the_voicemail_system.wav</audio> <audio type="file">${audio}/${lang}/mb_leave_msg_after_tone_two_minutes.wav</audio> <audio type="file">${audio}/${lang}/bi_beep.wav</audio>

 <event name="audio_end"><cmd>msg_create</cmd><cmd>goto record_message</cmd></event> <event name="0-9"><cmd>set collect ${digit}</cmd><cmd>goto collect_digits</cmd></event> <event name="#"><cmd>set collect</cmd><cmd>goto collect_digits</cmd></event> <event name="*"><cmd>set collect</cmd><cmd>goto collect_digits</cmd></event> </state>…

User defined account types may implement all kinds of media-related services like calling-card

Hidden settings

Web settings

Find the initial state

Description of the audio

State

State transition description

123

V1.0

2. H.323

3. SIP



6. Conclusion

7. Test


124

V1.0

Conclusion

• Voice over IP is a complex technology!

• However, has huge potentials– Using computer infrastructure instead of specialized

equipment– Integration with computer network makes things easier

(web browser, email-notification, etc.)

• SIP is getting mature– Most of the services known from PSTN are possible– Many new features that were not possible with PSTN

• snom offers phones and SIP proxy, media server

• VoIP technology can be integrated with any vendor– Quality is important

125

V1.0

2. H.323

3. SIP



6. Conclusion

7. Test


126

V1.0

Examination

sip:[email protected]

© 2003 snom technology Aktiengesellschaft

Written by:Dr. Christian StredickeVersion: 1.0

The author has made his best effort to prepare this document. The content is based upon latest information whenever possible. The author makes no representation or warranties of any kind with regard to the completeness or accuracy of the contents herein and accept no liability of any kind including but not limited to performance, merchantability, fitness for any particular purpose, or any losses or damages of any kind caused or alleged to be caused directly or indirectly from this document.

For more information, mail [email protected], Pascalstr. 10B, 10587 Berlin, Germany.

snom voip primer christian stredicke june 2003. v1.0 2 2. h.323 3. sip 4. snom technology suite 5....

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