MItel Technology Primer - SIP

7/29/2019 MItel Technology Primer - SIP

1/18

SIP Technical Overview

M I T E LM I T E L

Technology Primer

This primer is intended to explain some of theconcepts underlying SIP, its main characteristics that

make it a powerful protocol and the challenges facedby SIP especially in enterprise networks.This primer isintended to complement the first Mitel SIP Primer byproviding additional information on the protocol and itscapabilities. Both documents are complementary andcan be downloaded from the Mitel web site.

What is SIP?

SIP is a signaling protocol for controlling multi-mediasessions. It is used to establish user presence, locateusers (SIP enables mobility), as well as set up, modifyand tear down sessions.

Interaction with other protocols

From the previous definition, SIP is not a media or a

management protocol. In other words, SIP does notdefine new codecs, QoS nor is SIP voice specific. SIPrelies on other protocols such as RTP to transport userinformation (audio, video), DNS for address resolution,Diffserv / RSVP etc., for QoS, Radius / Diameter forAAA (Authentication,Authorization, Accounting), etc.SIP does not describe the audio and media componentsof a session; instead, it relies on a separate sessiondescription (SDP) carried in the body of SIP messages(INVITE and ACK).

The diagram on the next page shows the interaction

of these protocols. Of interest is the fact that SIP canbe implemented over UDP or TCP transport. SIPincorporates mechanisms to cover for packet losswith retransmissions based on timers.

The protocol of choice

From its initial standardization in 1999 by the IETF, Session Initiation Protocol (SIP) has rapidly

become the protocol of choice for the deployment of IP Telephony as evidenced by the public

service offering and announcements from MCI,Vonage, Packet 8, Telstra, SingTel, etc. SIP was alsoadopted by the Multiservice Switching Forum (MSF) for VoIP and VoATM trunking (SIP & SIP-T).

In addition, SIP received a strong boost in 2000 with its adoption for 3GPP networks (third

generation mobile). Even today SIP is used to offer such features as PTT (Push to Talk)

functionality and basic presence over 2.5G networks (1xRTT and GPRS).


2/182 | Mitel Technology Primer2 | Mitel Technology Primer

Technology Primer | SIP

M I T E L

Network

Transport

Link Layer &Physical Layer

Applications

Signaling MediaTransport

QoS Services/Utility

TCP UDP

IPv4, IPv6

SCTP

SIP

SDP

RTP RTCP

Media Coding

DNS DHCP

AAL5

ATM Ethernet

PPP

V.90, V.34

AAA

PPP

SONET/SDHWiFi, WiMax

Similarity to web protocols

SIP is modeled after HTTP in that it borrows the conceptsof URI, Schemes and Methods as implemented for theHTTP protocol. The Scheme used to identify the type ofresource is either SIP or SIPS (Secure SIP) and there areseveral access Methods (listed in the sections below).This design aspect is a major strength of SIP in that itcan readily integrate with other web infrastructures.

High level description

To better understand SIP, it is appropriate to describe thedifferent components of a SIP solution (SIP messages, SIPelements, SIP message flow). The fundamental tenets of

this protocol will be highlighted along with a comparisonto existing protocols (e.g., MGCP, H.323).

1. SIP elements

At a high level there are two types of SIP elements: UserAgents and Servers.

User Agents are endpoints in a SIP network: theyoriginate and terminate calls. Examples of User Agents(UA) include: SIP phones (hard sets), laptops or PDA witha SIP client (e.g., softphone), Media gateway (e.g. T1/E1gateway), access gateway (e.g., FAX gateway),conferencing systems, etc. All these devices also initiateand terminate the media session (voice, video, FAX, etc.).

A UA is itself comprised of two entities (software):

UAC (initiates call by sending INVITE with E.164 orURI dialing) UAS (receives call requests). More on SIP messages

and addressing to follow.

There are several types of servers in a SIP networkincluding Proxy server, Redirect server and SIPregistrar.

A Proxy server performs signaling and relay. In otherwords, it determines where to send signaling messagesand forward requests on behalf of the UA. To do so,it consults databases (DNS, location servers, etc.).

It is important to remember that Proxy servers haveno media capabilities; they are in the control path only.Proxy servers must pass unrecognized SIP messages

Interaction with other protocols


3/18Mitel Technology Primer | 3


M I T E L

through unchanged.Thus new features do not require

changes to proxy servers used in an infrastructure.This principle enables new features to be deployed in anetwork by only upgrading the end devices.

The routing function can be configured (programmed)according to user preferences, type of call (e.g., 911),least-GW-cost, or other criteria. Note that the proxyserver is not the only place where service can beprogrammed. In fact, service programmability can residein end-devices as well, such as for visual caller ID,distinctive ringing or possible Call Forwarding. Proxyservers can try several destinations sequentially or inparallel, this capability called forking enables multiple

devices to be associated with the same address.

There are three types of Proxy servers according to thetype of state information they keep: 1) a stateless proxykeeps no state, 2) a transaction stateful proxy only keepsstate on pending transactions, while, 3) a call statefulproxy keeps state for the entire duration of a SIP session.

Most implementations are stateful proxy-based as this isuseful for implementing such services as forward on noreply and also to implement forking. Stateless proxiesare easier to scale (especially under heavy load

scenarios) and can act as an application-layer loaddistributor (used in the core of a network). Redundancydesigns are easier to achieve with stateless proxies.

A SIP registrar accepts registration requests from users

(e.g., I am now at 192.168.0.10) and maintains userlocation information in a database. Mobility is thusachieved by the use of a REGISTER message (from UA)and by keeping a location database updated.

Redirect servers are servers that redirect SIP requeststo another device. A redirect server responds to therequest with the address to which the request shouldbe redirected to (e.g., a request for [email protected] canbe redirected to [email protected]).

SIP does not specify any implementation models for example, all above servers can reside on the same

hardware platform. The underlying OS can be Windows,Solaris, Linux or any embedded real time OS (QNX,VxWorks, MontaVista Linux, etc.). For example, VOCAL isan open-source VoIP software from Vovida.org. VOCALsoftware suite is a robust implementation of the SIPprotocol and its various entities and is used widely.

It is important to note that the above servers (proxy,redirect and registrar) are all optional SIP components.In fact, a UA may issue an INVITE directly to a targetedendpoint and many telephony features may beimplemented directly on the UA.The SIP model is

based on intelligent endpoints that can act withoutother intelligence from the network infrastructure (referto section below on peer-to-peer vs. centralized model).

SIP User

Agent

Registrar Redirect Proxy

1REGISTER

I am signing up2

INVITEConnect me [email protected]

5INVITE

Initate call tonic@home

66Media

SIP Servers

3Where is [email protected]?

4REDIRECTHe moved,

try him @home.com

[email protected] [email protected]

When a call comes in,apop-up window lets youknow who's calling.

Lastname, Firs t(123-4567)

People toCa ll

F rie nds(3 o f16a re on l ine )

79 People

Entera NameorPhoneNum ber...

MissedCalls

In the Offi ce

Imin ameeting

Line1001

Line1002

2xToda y4 :01p

Today 3 :11p

Today 2 :08p

2xToda y1 :37p

Today12: 59p

Henderson, Frank(234-5678)

Unknown (234-5678)

Lee, Bill(2342)

Jones, Ralph(5411)

Wilson,Pamela(2454)

11 NewIte ms

Lastname, First (123-4567)

Peopleto Call

Friends(3 of16 areonline)

ProjectTeam (1 of4 online)

Brown, Bill (Available)

Davidson, Karen (Busy)

Quick List 79 People

File Edit View Favorites Tools Help

CallEntera NameorPhoneNumber...

Line1001

Line1002

Example of User Mobility Using Register and Redirect Messages


4/184 | Mitel Technology Primer


M I T E L

2. SIP messages

There are two types of SIP messages: SIP requests(also called METHODs the same way as GET, PUT,DELETE, POST are METHODs for HTTP) and SIPRESPONSEs (shown in the tables below).

SIP requests (as defined in RFC3261) include thefollowing core METHODS:

INVITE to initiate a sessionRe-INVITE if, during a call, either party wants tochange the media; for example to open a video channelACK to confirm session establishment and can only be

used with INVITEBYE terminates sessionsCANCEL to cancel a pending INVITEOPTIONS for capability inquiryREGISTER to bind a permanent address tocurrent location

Other SIP method extensions are defined in differentRFCs such as:

SUBSCRIBE to subscribe to a service state change.Used for presence (subscribe to an event and receivenotification), call-back (when other party becomes

available), voice mail notification, any event that can be

associated with a trigger (e.g., stock quotes, etc.)NOTIFY notify a change of service state (e.g., newvoice message).Works in parallel with SUBSCRIBEMESSAGE for Instant Messaging (user to usermessaging). MESSAGE requests carry the content in theform of MIME body partsREFER call transferPUBLISH publication of presence information toa server

SIP is designed so that UAs can discover and negotiatetheir capabilities including what Methods are supported.Another aspect of negotiating capabilities include codec

support, handled by the SDP protocol. One UA in thesession generates an SDP message that includes (amongother information) all codecs the offerer wishes to use.The answer will indicate whether the stream is acceptedor not, along with the codecs that will be used and theIP addresses and ports that the recipient wants to use toreceive media.

SIP responses use a numerical code (borrowed fromHTTP response code, e.g., 404 Not Found) and a reasonphrase (see table below).

Code Type Description

1XX Information Request received continuing to process the request.

Example: 100 trying, 180 ringing

2XX Success The action was successfully received, understood and acceptedExample: 200 OK

3XX Redirection Further action must be taken to complete the request

Example: 301 Moved Permanently, 302 Moved Temporaily

4XX Client error Request contains bad syntaxExample: 400 Bad Request, 401 Unauthorized

5XX Server error Request cannot be fulfilled at this serverExample: 500 server Internal Error

6XX Global failure Request is invalid at any serverExample: 600 Busy Everwhere




M I T E L

3. SIP addressing

Because it is IP based, SIP provides users with

globally reachable addresses. These addresses (URI)use the same format as an email address: user@domain,(e.g., [email protected] or [email protected]). Users can haveany number of SIP URIs with different providers that allreach the same device. Instead of SIP URIs, users can beidentified also by telephone numbers, expressed astel URIs such as tel: +1-925-242-4321. Calls withthese numbers are then either routed to an Internettelephony gateway or translated back into SIP URIsvia the ENUM mechanism.

ENUM descriptionThe fundamental problem ENUM tries to solve is the

mapping between a standard telephone number anda SIP URI.

In enterprise networks today, this problem is addressedusing vendor proprietary implementations. These includerouting tables (in gateways, proxies, etc.) to translatethe dial strings to a host name to set up a call. ENUMis a better solution (especially for public VoIP service)because it solves inter-domain call routing based on atelephone number. In fact, until ENUM, there had beenno practical solution to the problem of call setup acrossthese domain boundaries.

The ENUM solution consists of a DNS-basedarchitecture and protocol to map dialed numbersto SIP URIs. In addition to providing the SIP URI,

ENUM can also provide such information as emailaddress, cell phone, VPIM information and FAX number.The advantage of using DNS is that it can be delegatedand it is scalable. In fact, each digit can be a definableDNS zone and zones can be delegated.

From a users perspective, ENUM is a transparentprocess. The ENUM logic and DNS resolution are carriedout in the background by ENUM-enabled devices, proxyservers or gateways.

After a user dials a phone number (e.g., 1-925-242-4321)the number is translated into a Fully QualifiedDomain Name (FQDN) that can be used by the DNS.For example, the above number can be translated into

1.2.3.4.2.4.2.5.2.9.1.e164.arpa. This FQDN is queriedfor NAPTR Resource Records. These records definethe services that can be associated with a particulartelephone number in ENUM, including SIP VoIP, fax,email, instant messaging, personal web pages, etc. Inthis case, the SIP phone or proxy would parse the NAPTRrecords looking for the service field that contained SIP.It would ignore all other records (mail to,tel, etc.)and then issue a SIP INVITE message to:sip:[email protected] in order to connect the call.

The example depicted below shows how ENUM

can operate between two SIP phones.The ENUMresolution service is invoked from a SIP phone thatissues a DNS query after the user dials a phone number(e.g., 1-925-242-4321). The information obtained fromthe NAPTR records is used to establish the call. In thecase of an analog phone, the ENUM service can beimplemented in the media gateway.

Proxy Server1

Query1.2.3.4.2.4.2.5.9.1.

e164.arpa

2Response

SIP:[email protected]

3INVITE

DSN

Server

INVITE

User dials1-925-242-4321

Proxy Server

Proxy Server

ENUM Description


6/18

InboundProxy Server

OutboundProxy Server

9180 Ringing

10180 Ringing

8180 Ringing

User Agent A User Agent B

When the phone is answered, the called UA sends a finalresponse with the media channels that it can support.Both parties agree on a media channel and the callerUA sends an ACK to the called UA. RTP streams canflow between devices.

6 | Mitel Technology Primer


M I T E L

4. Examples of SIP message flow

An example of a SIP message flow is shown at the right.To make a phone call for example, a SIP UA sends anINVITE request. In the message body, the UA specifiesthe type of media available. The outbound (receiving)Proxy server routes the request across the network untilit reaches its destination (multiple proxies can beinvolved).


InboundProxy Server

DNSServer

LocationServer

2100 Trying

4100 Trying

5LS Query: B

6Response:

sip:[email protected]

User Agent A

1INVITE

Contact:ASDP A

3INVITE

Contact: ASDP A

7INVITE

Contact: ASDP A

User Agent B

When the called party receives the INVITE request, itdetermines if it can accept the call in which case, it willring the phone and sends a provisional response back tothe caller (to indicate that the phone is ringing).

InboundProxy Server


12200 OK

Contact: BSDP B

User Agent A

13200 OK

Contact: BSDP B

14ACK

Media (RTP)

11200 OK

Contact: BSDP B

User Agent A User Agent B

Diagrams above borrowed with modifications

from Henry Sinnreich & Alan Johnston.




M I T E L

5. SIP and other protocols

An important difference between SIP and other protocolsis the fact that SIP endpoints can communicate directly.In other words, two SIP sets do not require any resourcesto establish a peer-to-peer communication, much in thesame way that two PCs can exchange a file (e.g., FTPclient / server) without any other devices. This capabilityis in contrast with stimulus based VoIP protocols such asMGCP that require intelligence to be located in thenetwork (Media Gateway Controller) for device control.Stimulus based protocols (e.g., MGCP, Megaco / H.248,PacketCable / NCS) have been deployed in large scalepublic networks for hosted IP Telephony (e.g., GoBeam /

Covad, Tiscali, Equant, etc.).The majority of enterprisenetworks deploying VoIP today also use some type ofproprietary stimulus based protocol.

MGCP and SIP can co-exist in VoIP networks, theycan especially be complementary in an environmentwith multiple softswitches (CA / MGC). This scenariodepicted below consists of using MGCP to control trunkgateways, low-end VoIP sets and IAD to deliver CLASSfeatures / services (but not advanced capabilities suchas presence, or video). SIP (or SIP-T) is then usedbetween Call Agent / MGC.

SIP is also superior to H.323 in many respects. First, it isflexible in that it can be implemented over TCP, UDP orSCTP and is not restricted to telephony only applications.Second, H.323 protocol structure is inherently muchmore complex, hence more difficult to implement.Third, SIP is inherently more extensible due to itsHTTP-like method / tags / MIME approach. SIP messagestructure (textual encoding) makes it easier to implementand add new functionality than H.323 that uses theITUs ASN.1 encoding standard instead of text. Lastly,SIP servers can be stateless (thus easier to scale) andSIP servers can ignore unknown headers whereascompatibility is required to operate; for example an

H.323v3 end-device on an existing H.323 infrastructure.

MGCPMGCP

Gateway Gateway

SoftSwitch SoftSwitch

SIP

RTP

SIP and other protocols




M I T E L

Applications

VoIPInfrastructure

PacketInfrastructure

PS/CA

MG

RoutingRouting RoutingTransmission

ASP

VoIP SP

NSP

AS

Business EntityServices

MS

Legend:AS: Applications Server CA: Call Agent (MGCP model)PS: Proxy Server MG: Media Gateway (e.g., Nuera)

MS: Media Server e.g., Convedia)

6. Protocol highlights and summary

In summary, some of the fundamental tenets ofthe SIP protocol are:

IP based protocol uses IP addressing End-to-end protocol messages make it to the other

end unaltered Unbundling of network transport from services ASP

can augment service offering Unbundling of services and applications quickly add

new applications No service intelligence in the network network has

routing knowledge and forwarding capabilities

No state knowledge or service logic in the network further unbundling between network and service Call and state intelligence resides in end devices

easy to scale total solution Intelligent endpoints can communicate without any

other resources

Client server based protocol Textual encoding easy to implement

and troubleshoot Multimedia can be used for voice, video,

gaming, IM, etc.

While some of the above points also characterize thePSTN and its underlying protocols (e.g., SS7), SIPenables a new level of autonomy between servicesand applications, furthermore, services may be offeredby different providers (ASP). In other words, a usercan subscribe to more than one provider for signaling

(a side benefit is to gain back service in case of failure)to another provider for connectivity to legacy networks,while subscribing to an ASP for an IVR service, forexample. Most importantly, adding a new applicationor functionality is a trivial exercise when compared toadding the same functionality to a legacy PSTN network.

SIP Enables a new Business Model Between Service Providers




M I T E L

7. SIP and third-party control

SIP is designed so that two entities (users / services)can jointly establish a communication. Some serviceshowever require a third party involved to establish thecommunication channel. This is the case for example ofclick-to-dial (where a controller establishes a call on thecallers behalf), IVR (where the AS determines where tosend the call after initial input from the caller) or prepaidcalling (where the caller initially enters information intoa controller). Third-party Control refers to the ability fora device that is not one of the ends of the SIP signalingto affect a SIP dialog. Third-party Call Control is not aSIP extension but a clever mechanism that allows a

controller (UA) to independently exchange signaling withtwo parties (A & B) and convinces them to send media toeach other. In fact, the two parties believe that they arein session with the controller but effectively they aresending media to each other.

Third-party control, also called centralized modelbecause it requires a central point of control, may not bedesirable in some environments. An alternative approachto perform call control is based on a peer-to-peer model(distributed) which uses SIP REFER and Replaces Header.The Replaces Header is used to logically replace anexisting SIP dialog with a new SIP dialog. One useof the Replaces Header is to replace one participant withanother and is frequently used in combination with theREFER method, for example to retrieve a parked call.

8. Presence, IM and SIP

SIP enables basic messaging between two parties(using the SIP MESSAGE method described above.The MESSAGE method provides pager-mode messagingwhere messages sent are independent of each other(no concept of a session) similar to a two-way pagerservice. The request may traverse a set of SIP proxies,using a variety of transports, before reaching itsdestination. This mode, suitable for short messagesor broadcast information (e.g., server re-boot in twominutes), has been criticized for its relative highoverhead and lack of true IM functions.

An IETF working group (SIMPLE or SIP for InstantMessaging and Presence Leveraging Extensions) isfocused on the application of the Session InitiationProtocol to Instant Messaging and Presence (IMP).One of the main benefits of this effort is the recognitionof the distinction between presence and messaging andto standardize the protocol to enable interoperabilitywith different vendors.

A second mode (session-mode) was introduced toprovide ordering security.This mode was designednot to burden the SIP signaling network by workingdirectly between the endpoints. There is, however,

more complexity (e.g., a new protocol: MSRP mustbe implemented in end devices) to contend with.It is important to note that the IETF has also blessedother competing specifications for Presence andInstant Messaging, notably XMPP (jabber).

Lastname,Firs t (123-4567)

People to Call


ProjectTeam(1of4online)

Brown, Bill(Available)Davidson, Karen (Busy)

79 People



AutoAnswer

Call ForwardProfile

Contextsensitiveinstructionaltextdisplayed here...

IntheOffice

Im in a meeting

Line1001

Line1002

2xToday4:01p

Today3:11p

Today2:08p

2xToday1:37p

Today12:59p

Henderson, Frank(234-5678)

Unknown(234-5678)

Lee, Bill (2342)

J ones , Ralph (5411)

Wilson, Pamela (2454)

11 NewItems

Lastname, First (123-4567)

Peopleto Call


ProjectTeam (1 of4 online)

Brown, Bill (Available)

Davidson, Karen (Busy)

Quick List 79 People



Line1001

Line1002

Message

200 OK

SIP MESSAGE method is used to sendinstant messages, where each message isindependent of any other messsage

Application / Feature Server

Presence, IM and SIP

SIP and third-party control




M I T E L

9. SIP-based deployment

This section provides three examples of SIP deployment:one in public networks, one in enterprise networks andone in private-public applications.

1. Augmenting existing Class 5 switches with SIPTraditional service providers while wanting to offer newservices leveraging the power of SIP also place a greatemphasis on preserving their investment in TDMinfrastructure. A SIP-based solution should co-exist withthe existing Class 5 switches while allowing serviceproviders to generate a new stream of revenues fromexisting and new subscribers.

This is the premise behind products such as the Mitel3600 Integrated Communications Platform (ICP) server(or product offering from other vendors) that as a SIPSmall Business Feature Server, it can connect to legacyswitches and deliver a whole range of advanced IPservices. Some of these services include web portal,mobility, teleworking and self-provisioning.

Gateway

PRI

Class 5 switch

Mitel 3600 ICP Server

PSTN

BroadbandNetwork

Augmenting Existing Class 5 Networks with SIP




M I T E L

2. Migrating to next generationSIP-based messaging systemsMany enterprises are faced with the upgrade oftheir VMS that reach their end of life cycle and look toenable new functionality such as unified messaging.Selecting a SIP-based platform is a difficult choice. In

fact, the platform has to integrate with the legacy PBXand legacy VMS (in the case of distributed networks).This implies that the SIP-based Media Server mustaccommodate analog or digital connectivity to PBXs(in addition to IP) and support message exchange using

VPIM and AMIS. This capability exists today on the MitelNuPoint Messenger Model 70 IP (offerings from othervendors too). The Mitel solution supports native SIP inaddition to the integration to a dozen traditional PBXs.

Using a flexible SIP media server, such as the NuPointMessenger Model 70 IP, enterprises can smoothlymigrate to a SIP infrastructure and accommodatedistributed as well as centralized messaging architecturesas depicted above.

Traditional PBX

digital

NuPoint UM - SIP

NuPoint UM - SIP

TraditionalCentrex Service

SMDI

IP/ VPIM

T1

IP / VPIM

IP/ VPIM

NuPoint UM - SIP

Site #1

Layer 2

Site #2

Site #3

SIP IPBX

IP WAN

PSTN

SIP-Based Unified Messaging Deployment




M I T E L

3. Merging IPBX with public VoIP infrastructures

using SIPCustomers using a hosted or Centrex service, havetraditionally had limited access to advanced applicationssuch as teleworking, unified messaging, contact centerapplications, conferencing and collaboration solutions.On the other hand, customers deploying IPBX face manychallenges in deploying multi-site networking including:(1) site-to-site connectivity over IP, (2) managing PSTNconnectivity, (3) managing billing (one bill for all sites),(3) configuring dialing plans, (4) call routing, etc.

IPC2 is the SIP answer from Marconi and Mitel (offeringavailable from other vendors) to enable service providersto leverage IPBX for advanced features at the customerpremises while offering business trunking and VPNservices using a soft-switch architecture (other servicesare also enabled in this architecture). Business trunkingand VPN services enable customers to control their IPBXwhile billing, call routing and site to site connectivity arehandled by the Service Provider.

Head Office Regional Office

Remote Users

PSTN

Legacy PBX

Mitel 3300 ICP

Video

ApplicationServer

OtherApplication

Voicemail

Mitel 3300 ICP

Branch Office

IP-VPN

NetworkGateway

SoftSwitchMedia

Firewall

Merging IPBX with Public VoIP Infrastructures Using SIP




M I T E L

10. Centralized vs. distributed

deployment models B2BUAOne of the fundamental premises behind SIP isits distributed nature and the fact that calls areend-to-end. SIP servers as noted throughout are optional.Several vendors, deviating from the previous model, offera centralized architecture also referred to as back-to-back User Agent implementation (B2BUA). Sucharchitecture consists of using the SIP server as amediation device for all calls. In other words, a B2BUAserver appears just as another SIP endpoint and canmodify the message (as depicted below).

With a B2BUA implementation, it can be easier to offerPBX-like features, manage calls end-to-end (CDR, billing,etc.), implement and enforce policies (CoS, CoR, etc.) andaddress NAT issues (described in the next section).A market segment where this solution is well received isSmall and Medium size businesses (




M I T E L

If the call is disconnected can the PSAP contact

the initiator of the emergency call? Where to?Who provides this information? Who provides Caller Identity Validation? If there is no intelligence in the network, there may

be no VoIP SP involved and ISPs do not track whattype of packets are sent. How will the user contactthe appropriate PSAP?

In the above scenario is the ISP responsible toguarantee call completion?

In the long-term, users may not have E.164 numbersWhat URI is used? Is it ubiquitous?

How to determine location information?Who maintains location information?

Will it handle mobility?

Other issues, not SIP related, include:

Mobile and traditional analog phones do not have apower supply whereas most SIP desk phone will stopoperating under power loss

Who should pinpoint the exact location of user in aWiFi hotzone (and how)? How is this informationconveyed to the user? To the PSAP?

There are several technologies available that can

come to the rescue (e.g., DHCP tagging and extensionsto identify location, 802.11 triangulation, GPS, etc.).In general, there is an agreement that a SIP-based VoIPoffering should proceed ahead as these issues are beingaddressed in various organizations (NENA,APCO,CGALIES, ETSI, etc.).

To support CALEA (Communications Assistance for LawEnforcement), a telecommunications carrier must ensurethat its equipment, facilities, or services are capable ofisolating and enabling the government to intercept allwire and electronic communications and providingaccess to call-identifying information. Using a pure SIPpacket-based infrastructure however introduces newchallenges in that there is no standard handoverinterface for packet-based networks into an LEAcollection node (Law Enforcement Agency). Furthermoresubscribers may not be identified using a fixed directorynumber but using SIP URL.

13. Other SIP challenges

SIP has been proven in deployments exceeding 200,000users (Free World Dialup, Vonage, SIP.edu initiative, etc.).Complex issues remain including reliability, featurerichness, security, privacy and NAT traversal.

Reliability issues are mostly evident in implementationsof stateful proxies during failure of the primary proxyserver. Failure detection and switchover can take a longtime especially if SIP over UDP is implemented (ratherthan TCP).

Lack of feature support is not a SIP limitation, it is rather

a result of a vendors decision to offer a limited numberof features, but interoperable. There is ongoing effort thesupport of a large number of features in SIP (SIMPLE).

Securing a protocol like SIP is very complex. Issuesinclude authentication, authorization, message integrityand privacy. These security issues are being addressedby extensions to the protocol. SIPS, similar to HTTPS,mandates the use of a secure transport protocol, suchas TLS, between trusted entities. S/MIME (RFC 1847) isfor end-to-end message authentication and validation,and encryption of message bodies. These extensions arenot widely implemented yet.

NAT traversal is a relatively complex issue.The challengeis getting media sessions to pass through NAT deviceswhen the caller is trying to reach a party behind aNAT device. Several solutions have been proposedsuch as STUN and TURN. These solutions have theirown drawbacks. In the case of STUN it does not workacross all types of NAT devices (more specificallysymmetric NAT). Another approach is the use ofApplication Level Gateways (ALG) that are specializedfirewalls that understand specific IP protocols such asSIP, and dynamically open those ports needed by the

application leaving all others securely closed. Upgradinga firewall with ALG functionality can be expensive, asthe firewall needs to have intimate knowledge ofprotocol implementations. This would also imply thatamending a protocol or adding new protocols requiresinfrastructure change. So much for unbundlinginfrastructure from applications.




M I T E L

In conclusion

SIP deployment in the enterpriseThe main appeal of IPT is to enable new applicationsincluding convergence to the user and to lower the totalcosts of operating a voice network. The main appeal ofSIP is in its standards based approach that ultimatelyoffers customers even better ROI (Return on Investment)by offering a wider selection of appliances, servers,services, etc., (side benefit of competition). The ROI isalso achieved by not locking customers into a proprietaryprotocol that will prove expensive to migrate from.

To some extent, the success of SIP in public networks

contrasts with a milder reception of SIP into enterprisemarkets where vendor protocols (Cisco / SCCP, Mitel /Minet, Nortel / UniStim, Avaya / CCMS+H323, etc.) aremostly being deployed.

It is important to note that a basic level ofinteroperability can be easily achieved betweendifferent vendors (Mitel, Cisco, Polycom, BroadsoftSIP proxy server, etc.). In fact, Mitel SIP phones canbe added to a SIP infrastructure with Cisco SIP setsalongside with other sets from Polycom. More advancedfeatures however (IM, security, etc.) or private SIPextensions are not always supported across all vendors

and some integration work is required for more complexsettings (e.g., contact centers, unified messaging andnotification, VPIM to legacy VMS, etc.).

While integration is not an issue for service providers orlarge enterprises, it can represent a substantial effort formedium size organizations (




M I T E L

Acronyms

1xRTT Single Carrier Radio Transmission Technology3G Third Generation (wireless)3GPP 3G Partnership Project (UMTS)AAA Authentication, Authorization and Accounting (IETF)AG Access GatewayAPCO Association of Public-Safety Communications OfficialsAS Application ServerASP Application Server ProviderCDR Call Detail RecordingCGALIES Group on Access to Location Information by Emergency SvsCLASS Custom Local Area Subscriber Services,

aka Custom Calling features

COR Class of RestrictionCOS Class of ServiceDTMF Dual Tone/Multiple FrequencyENUM E.164 Numbering in DNS (IETF RFC 2916)ETSI European Telecommunications Standards InstituteFQDN Fully Qualified Domain NameGK GatekeeperGPRS General Packet Radio ServiceIETF Internet Engineering Task ForceIMP Instant Messaging and PresenceISP Internet Service ProviderIVR Interactive Voice ResponseJAIN Java Application Interface Network

LDAP Lightweight Directory Access Protocol (IETF)MG Media GatewayMGCP Media Gateway Control Protocol (IETF, ITU-T J.162)MPLS Multi-Protocol Label SwitchingMS Media ServerMSRP Message Session Relay ProtocolNAPTR Naming Authority PointerNCS Network Call/Control Signaling (PacketCable MGCP)NENA National Emergency Number AssociationNGN Next Generation NetworkPA Presence AgentPUA Presence User Agent

PBX Private Branch eXchangePOTS Plain Old Telephone ServicePSTN Public Switched Telephone NetworkQoS Quality of ServiceRFC Request For Comment (IETF)


17/18


M I T E L

ROI Return on Investment

RTCP Real Time Transport Control Protocol (IETF)RTP Real Time Transport Protocol (IETF RFC 1889)SCTP Stream Control Transmission ProtocolSDP Session Description Protocol (IETF RFC 2327)SIMPLE SIP Instant Messaging and Presence Leveraging ExtensionsSIP Session Initiation Protocol (IETF)SIP-T SIP For Telephony (IETF)SNMP Simple network management protocolSP Service ProviderSRV Server location records extension to DNSTDM Time Division MultiplexingTRIP Telephony Routing over IP (IETF RFC 2871)UA User Agent

URI Uniform Resource IndicatorVoIP Voice over IPVPIM Voice Profile for Internet MailXMPP Extensible Messaging and Presence Protocol

Acknowledgements

The author would like to thank the many colleagues throughout industry andacademia in the IETF SIP, SIMPLE and SIPPING working groups that develop IPcommunications technology.

The information in this document is believed to be accurate at the time of

publication. Contact Mitel directly for updated information or for more details.

www.mitel.com

North America

(613) 592 2122

1 800 648 3579

Latin America

(613) 592 2122

1 800 648 3579

UK

Tel: +44 (0)1291 430000

Fax: +44 (0)1291 430400

France

Tel: +33 (0)1 61 37 00 90

Fax: +33 (0)1 61 37 00 99

THIS DOCUMENT IS PROVIDED TO YOU FOR INFORMATIONAL PURPOSES ONLY. The information furnished in this document, believed by Mitel to be accurate as of thedate of its publication,is subject to change without notice.Mitel assumes no responsibility for any errors or omissions in this document and shall have no obligation toyou as a result of having made this document available to you or based upon the information it contains.

M MITEL (design) is a registered trademark of Mitel Networks Corporation. All other products and services are the registered trademarks of their r espective holders.

Copyright 2004, Mitel Networks Corporation. All Rights Reserved. GD 7815 PN 51008441RA-EN

Benelux

Tel: +31 (0)30 85 00 030

Fax: +31 (0)30 85 00 031

Italy

Tel:+39 02 2130231

Fax: +39 02 21302333

Germany, Switzerland,Austria

Tel: +49 (0)211 5206480

Fax: +49 (0)211 52064899

Portugal and Spain

Tel: +34 91 350 66 33

Fax: +34 91 350 70 14

Middle East

Tel: +971 4 3916721

Fax: +971 4 3915288

South Africa

Tel:+27 82 559 8688

Fax: +27 11 784 6916

Asia-Pacific

Tel: +852 2508 9780

Fax: +852 2508 9232


18/18

Reference Material

The following is a list of reference materials on SIP.

Books SIP Demystified. Gonzalo Camarillo, McGraw-Hill Telecom, ISBN 0-07-137340-

3

Internet Communications Using SIP: Delivering VoIP and MultimediaServices with Session Initiation Protocol, Sinnreich, Henry & Alan B.

Johnston, John Weily & Sons, Inc., New York (c) 2001, ISBN 0-471-41399-2

SIP: Understanding the Session Initiation Protocol, Alan B. Johnston,Artech House, Boston, London, January 2001, ISBN 1-58053-168-7

Web-based Information

Session Initiation Protocol (SIP) http://www.cs.columbia.edu/sip The Internet Engineering Task Force (IETF) Web site http://www.ietf.org SIP Center http://www.sipcenter.com VOIP Wiki - a reference guide to all things VOIP www.voip-info.org/wiki-SIP The SIP Forum is an industry organization with members from the leading SIP

technology companies. Its mission is to advance the adoption of products andservices based on SIP. www.sipforum.com

White PapersThe following white papers are available from Mitel OnLine.

Examining the Value of SIP in the EnterpriseWhite Paper (PDF 238KB)

SIP: Enabling Your Business to Leverage the Power of the Internet

Customer Brief (PDF 273KB)

SIP Interoperability List (PDF 77KB)

SIP Technology Primer - Technical Overview

(PDF 2.7MB)

SIP Technology Primer - Value Proposition

(PDF 490KB)

5215/5220 IP Phones (SIP functionality)

Data Sheet - English (PDF 545KB)
http://www.cs.columbia.edu/siphttp://www.ietf.org/http://www.sipcenter.com/http://www.voip-info.org/wiki-SIPhttp://www.sipforum.com/http://www.sipforum.com/http://www.voip-info.org/wiki-SIPhttp://www.sipcenter.com/http://www.ietf.org/http://www.cs.columbia.edu/sip

Documents

MItel Technology Primer - SIP