7815-Sip Tech Primer · Technology Primer This primer is intended to explain some of the concepts underlying SIP, its main characteristics that ... WiFi, WiMax SONET/SDH

SIP Technical Overview

M I T E LM I T E L

Technology Primer

This primer is intended to explain some of the concepts underlying SIP, its main characteristics thatmake it a powerful protocol and the challenges faced by 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 protocolsFrom the previous definition, SIP is not a media or amanagement 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 for AAA (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 can be implemented over UDP or TCP transport. SIPincorporates mechanisms to cover for packet loss with retransmissions based on timers.

The protocol of choiceFrom 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 also

adopted 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).

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Technology Primer | SIP

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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 protocolsSIP 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 descriptionTo better understand SIP, it is appropriate to describe thedifferent components of a SIP solution (SIP messages, SIPelements, SIP message flow). The fundamental tenets ofthis protocol will be highlighted along with a comparisonto existing protocols (e.g., MGCP, H.323).

1. SIP elementsAt 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 or

URI 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 have no media capabilities; they are in the control path only.Proxy servers must pass unrecognized SIP messages

Interaction with other protocols

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through unchanged. Thus new features do not requirechanges 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 multipledevices 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 loadscenarios) 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 requests to another device. A redirect server responds to therequest with the address to which the request should be 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 samehardware 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 without other intelligence from the network infrastructure (referto section below on peer-to-peer vs. centralized model).

SIP UserAgent

Registrar Redirect Proxy

1REGISTER

I am signing up2

INVITEConnect me [email protected]

5INVITE

Initate call tonic@home

66Media

SIP Servers3

Where is [email protected]?

4REDIRECTHe moved,

try him @home.com

[email protected] [email protected]

When a call comes in,a pop-up window lets you know who's calling.

Lastname, First (123-4567)

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Friends (3 of 16 are online)

79 People

Enter a Name or Phone Number...

Missed Calls

In the Office

Iím in a meeting

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2x Today 4:01p

Today 3:11p

Today 2:08p

2x Today 1:37p

Today 12:59p

Henderson, Frank (234-5678)

Unknown (234-5678)

Lee, Bill (2342)

Jones, Ralph (5411)

Wilson, Pamela (2454)

11 New Items


People to Call


Project Team (1 of 4 online)

Brown, Bill (Available)

Davidson, Karen (Busy)

Quick List 79 People

File Edit View Favorites Tools Help

CallEnter a Name or Phone Number...

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Example of User Mobility Using Register and Redirect Messages

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2. SIP messagesThere 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 session Re-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 beused with INVITEBYE – terminates sessionsCANCEL – to cancel a pending INVITEOPTIONS – for capability inquiryREGISTER – to bind a permanent address to current 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 beassociated 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 to a server

SIP is designed so that UAs can discover and negotiatetheir capabilities including what Methods are supported.Another aspect of negotiating capabilities include codecsupport, 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 accepted Example: 200 OK

3XX Redirection Further action must be taken to complete the requestExample: 301 Moved Permanently, 302 Moved Temporaily

4XX Client error Request contains bad syntax Example: 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



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3. SIP addressingBecause 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 as “tel” URIs such as tel: +1-925-242-4321. Calls withthese numbers are then either routed to an Internettelephony gateway or translated back into SIP URIs via the ENUM mechanism.

ENUM descriptionThe fundamental problem ENUM tries to solve is themapping between a standard telephone number and a SIP URI.

In enterprise networks today, this problem is addressedusing vendor proprietary implementations. These includerouting tables (in gateways, proxies, etc.) to translate the dial strings to a host name to set up a call. ENUM is 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-based architecture and protocol to map dialed numbers to 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 user’s 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 Qualified Domain Name (FQDN) that can be used by the DNS.For example, the above number can be translated into1.2.3.4.2.4.2.5.2.9.1.e164.arpa. This FQDN is queried for NAPTR Resource Records. These records define the 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 that issues 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

DSNServer

INVITE

User dials1-925-242-4321

Proxy Server

Proxy Server

ENUM Description

Inbound Proxy Server

Outbound Proxy 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 caller UA sends an ACK to the called UA. RTP streams can flow between devices.



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4. Examples of SIP message flowAn 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).



DNS Server

Location Server

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).



12200 OK

Contact: B SDP B

User Agent A

13200 OK

Contact: B SDP B

14ACK

Media (RTP)

11200 OK

Contact: B SDP B

User Agent A User Agent B

Diagrams above borrowed with modifications

from Henry Sinnreich & Alan Johnston.



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5. SIP and other protocolsAn 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, they can especially be complementary in an environment with 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 such as 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 its HTTP-like method / tags / MIME approach. SIP messagestructure (textual encoding) makes it easier to implementand add new functionality than H.323 that uses the ITU’s ASN.1 encoding standard instead of text. Lastly,SIP servers can be stateless (thus easier to scale) and SIP servers can ignore unknown headers whereascompatibility is required to operate; for example anH.323v3 end-device on an existing H.323 infrastructure.

MGCPMGCP

Gateway Gateway

SoftSwitch SoftSwitch

SIP

RTP

SIP and other protocols



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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 summaryIn summary, some of the fundamental tenets of the 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), SIP enables a new level of autonomy between services and applications, furthermore, services may be offered by different providers (ASP). In other words, a user can 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 application or 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



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7. SIP and third-party controlSIP 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 thecaller’s 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 for a device that is not one of the ends of the SIP signalingto affect a SIP dialog. Third-party Call Control is not a SIP extension but a clever mechanism that allows acontroller (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 model because 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 use of 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 messages or 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 designed not to burden the SIP signaling network by workingdirectly between the endpoints. There is, however,more complexity (e.g., a new protocol: MSRP must be implemented in end devices) to contend with.It is important to note that the IETF has also blessed other competing specifications for Presence and Instant Messaging, notably XMPP (jabber).


People to Call





79 People



Auto Answer

Call Forward Profile

Context sensitive instructional text displayed here...

In the Office

Iím in a meeting

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2x Today 4:01p

Today 3:11p

Today 2:08p

2x Today 1:37p

Today 12:59p

Henderson, Frank (234-5678)

Unknown (234-5678)

Lee, Bill (2342)

Jones, Ralph (5411)

Wilson, Pamela (2454)

11 New Items


People to Call





Quick List 79 People



Line 1001

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Message

200 OK

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

Application / Feature Server

Presence, IM and SIP

SIP and third-party control



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9. SIP-based deploymentThis 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

Broadband Network

Augmenting Existing Class 5 Networks with SIP



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2. Migrating to next generation SIP-based messaging systemsMany enterprises are faced with the upgrade of their 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. Infact, 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

Traditional Centrex 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



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3. Merging IPBX with public VoIP infrastructuresusing 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

Application Server

OtherApplication

Voicemail

Mitel 3300 ICP

Branch Office

IP-VPN

NetworkGateway

SoftSwitchMedia

Firewall

Merging IPBX with Public VoIP Infrastructures Using SIP



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10. Centralized vs. distributed deployment models – B2BUAOne of the fundamental premises behind SIP is its distributed nature and the fact that calls are end-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 (<5,000 users) wherecustomers prefer a central point of service for all usersand where policies, security, firewall and other servicescan be managed, enforced and terminated.

11. SIP and managementManaging traditional telephony relies on proprietaryvendor implementations to address such issues as faultmanagement, configuration management (adding auser), accounting (extracting CDRs), performance andsecurity (setting a COR or policy).

By disagregating the PBX (and the Class 5 switch), IPT(SIP included) enables the deployment of a standardsbased management framework. This framework includes protocols such as Radius for AAA (user policyconfiguration, accounting, etc.), LDAP for Directory,SNMP MIB for user and network devices, SNMP traps to collect alarms from multiple devices and FTP / TFTPservers to collect and retrieve metrics (RTCP, etc.).Furthermore, web tools (web browsers) can be used toview and configure the above information, lastly moresophisticated system / network managers (HP OV NNM)can be used to manage all above network elements (IP sets, trunk gateway, etc.).

While in theory, IPT can re-use the existing datamanagement framework, it is a challenging task for many enterprise network managers to integratedisparate network elements (media gateways, useragents, media servers, etc.) into their existingmanagement infrastructure. Service providers, on the other hand, are faced with many new challengessuch as Service Level Management (integration withmultiple network elements), user self-configuration(security issues) and unifying information from different devices (many devices generate CDRs).

12. CALEA and E911 / E112Delivering E911 service in SIP-based networks add alayer of complexity that is best understood when tryingto address some of the questions below:

• Users can move to a different house without callingthe carrier. Who is responsible for routing their callsto the appropriate PSAP? E.164 numbers may notreflect geographic area as a caller in California mayhave a NY number

• If the Proxy server must determine where to route theemergency call then where does it route the call if theuser is in a different state?

User Agent 1

Dialog #2:UAC Dialog initiated

Dialog #1:UAS Dialog initiated

SIP Server: B2BUA Implementation

1

4 2

3

User Agent 2

SIP-based B2BUA Deployment Model



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• 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 challengesSIP 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 rathera result of a vendor’s 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 addressed by extensions to the protocol. SIPS, similar to HTTPS,mandates the use of a secure transport protocol, such as 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 a NAT device. Several solutions have been proposed such as STUN and TURN. These solutions have their own 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 theapplication leaving all others securely closed. Upgradinga firewall with ALG functionality can be expensive, as the 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 conclusionSIP 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 networkscontrasts 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 of interoperability can be easily achieved between different vendors (Mitel, Cisco, Polycom, Broadsoft SIP proxy server, etc.). In fact, Mitel SIP phones can be 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 vendorsand 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 (<5,000 users). This could be one of many reasons why SIP is lagging in mediumsize enterprise networks. As SIP matures and moreinteroperability testing is conducted, SIP will become the dominant protocol in enterprise markets.

For more on SIP, go to:www.voip-info.org/wiki-SIPwww.cs.columbia.edu/sip/ www.cs.columbia.edu/sip/faq/ www.greycouncil.com/sipwg/ – SIP WG Supplemental Homepagewww.sipcenter.com/ – The SIP Centerwww1.cs.columbia.edu/~xiaotaow/sipc/ – UoColumbiaSIP User Agent (sipc)www.ubiquity.net/products/SIP/SIP_User_Agent.php– Ubiquity SIP user agentwww.sipforum.comwww.softarmor.com

For more information or to evaluate Mitel SIP solutions,please contact your local Mitel Sales and SystemsEngineer or visit our web site at http://www.mitel.com



M I T E L

Acronyms1xRTT 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” featuresCOR 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 NetworkLDAP 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 AgentPBX Private Branch eXchangePOTS Plain Old Telephone ServicePSTN Public Switched Telephone NetworkQoS Quality of ServiceRFC Request For Comment (IETF)


M I T E L

ROI Return on InvestmentRTCP Real Time Transport Control Protocol (IETF)RTP Real Time Transport Protocol (IETF RFC 1889)SCTP Stream Control Transmission Protocol SDP 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 AgentURI Uniform Resource IndicatorVoIP Voice over IPVPIM Voice Profile for Internet MailXMPP Extensible Messaging and Presence Protocol

AcknowledgementsThe 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 ofpublication. Contact Mitel directly for updated information or for more details.

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