direct sip

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Microsoft Lync Server 2010 Resource Kit Direct SIP

This chapter is part of the Microsoft Lync Server 2010 Resource Kit book that is currently being developed. Chapters will be available for download while this book is being completed. To help us improve it, we need your feedback. You can contact us at [email protected]. Please include the chapter name.

For information about the continuing release of chapters, check the DrRez blog at http://go.microsoft.com/fwlink/?LinkId=204593.


Contributors

Project Manager: Susan S. Bradley

Content Architect: Rui Maximo

Chapter Lead: Taimoor Husain

Technical Reviewers: Kam Toor, Ken Araujo

Lead Editor: Kelly Fuller Blue

Art Manager: Jim Bradley

Production Editor: Kelly Fuller Blue


Table of ContentsContributors................................................................................................................................................4

Introduction.................................................................................................................................................6

Requirements of Direct SIP from Lync Server..........................................................................................6

Scenarios.....................................................................................................................................................7

Inbound Calls to Lync Server....................................................................................................................7

Outbound Calls from Lync Server............................................................................................................8

PBX User Joining Lync Server Audio Conference...................................................................................10

Direct SIP with Media Bypass................................................................................................................11

Direct SIP with Simultaneous Ringing....................................................................................................12

Direct SIP Configuration............................................................................................................................14

Lync Server Configuration......................................................................................................................14

Cisco Unified Communications Manager Configuration........................................................................19

Dial Plan.....................................................................................................................................................23

Outbound from Lync Server..................................................................................................................24

Summary...................................................................................................................................................30


Introduction

When Microsoft® Lync™ Server 2010 communications software is deployed in an enterprise environment that already has telephony infrastructure, some level of interoperability is required. This allows users and devices on the existing infrastructure to communicate with Lync Server 2010 users and devices.

Within the telephony environment of an enterprise, it is not uncommon to have multiple communications systems deployed to accommodate the voice workload. Such a system is called a PBX (Private Branch Exchange). Some legacy Time Division Multiplexing (TDM) PBX systems can be upgraded to support IP connections. Such a system is called IP-PBX. Legacy systems and IP-PBX systems form the telecommunications infrastructure for the enterprise.

Direct SIP is referred to as a SIP trunk between an IP-PBX and Lync Server. Because SIP is a standards-based protocol, communications between endpoints on either the IP-PBX or the Lync Server call control plane can be provided by using direct SIP. It can also be used to enhance the existing telephony infrastructure by extending Microsoft Lync 2010 features such as dial-in audio conferencing and Response Groups to the IP-PBX.

Requirements of Direct SIP from Lync ServerIn Lync Server 2010, direct SIP is configured between a Microsoft Lync Server 2010, Mediation Server and an IP-PBX. The Lync Server 2010, Mediation Server connects both signaling and media to SIP peers (that is, gateways, Session Border Controllers (SBCs), proxies, IP-PBXs, and so on). In the case where media bypass is supported, the media traffic can be directly established between the Lync 2010 client and the SIP peer, bypassing the Microsoft Lync Server 2010, Mediation Server. Media bypass optimizes the media channel network path.

For a gateway, SBC, proxy, and IP-PBX to connect to the Lync Server 2010, Mediation Server, it must support certain requirements. The direct SIP peer must support the following:

SIP over Transmission Control Protocol (TCP) G.711 codec RFC2833 for dual-tone multifrequency (DTMF)

Codec selection and media flow considerations are covered later in this chapter.


ScenariosFigure 1 shows direct SIP connectivity between Lync Server and Cisco Unified Communications Manager (CUCM).

Figure 1. Direct SIP connection to CUCM

Inbound Calls to Lync ServerWhen a user who is using an IP-PBX phone that is connected to CUCM makes an inbound call to a Lync 2010 user, CUCM sends a SIP INVITE to the Mediation Server. The SIP INVITE contains the destination phone number in the TO: field. The Mediation Server applies the appropriate context to the header, and then routes the call to the callee’s Front End pool. The pool sends the INVITE to all of the callee’s registered endpoints. The following steps describe how the inbound call is routed; see figure 2.

1. IP-PBX phone signals CUCM to initiate a call.2. CUCM checks its routing table, and then determines that the phone number that has

been called is to be routed to the SIP trunk to the Mediation Server. CUCM send a SIP INVITE to the Mediation Server, where the called number is listed in the TO: box of the SIP header.

3. The Mediation Server acknowledges the INVITE.4. The Mediation Server applies the phone-context, based on the configured dial plan

for the SIP trunk, and then routes the INVITE to the associated Microsoft Lync Server 2010, Front End pool.

5. The Lync Server 2010, Front End pool uses the phone-context to normalize the incoming number, based on the matching dial plan. The Front End pool then performs a reverse number lookup (RNL) to find the SIP URI of the user. The Front End pool


determines the registered endpoint or endpoints of the called user, and then routes the SIP INVITE to those endpoints.

6. After the callee accepts the call, the media path is directly established between the Lync 2010 client, the Mediation Server, and CUCM to the IP phone. In the case where media bypass is configured, the media path would bypass the Mediation Server and directly connect from the Lync 2010 client to CUCM

Figure 2. Inbound call from IP-PBX phone to Lync 2010

Outbound Calls from Lync ServerWhen a Lync 2010 user calls an IP-PBX phone that is connected to the IP-PBX, the phone number is normalized by the Lync 2010 client. It uses the normalization rules that are defined in the voice policy that is assigned to the user. Lync 2010 sends the SIP INVITE to the user’s Front End pool. The Front End Server performs an RNL. Because it doesn’t find a match for the number, the Front End Server sends the number to the outbound routing component of the Front End. Outbound routing finds a route that matches the prefix of the phone number that was called. This route points to the IP-PBX. The Front End then routes the INVITE to the Mediation Server that is associated with the IP-PBX. The Mediation Server sends the call through the direct SIP (also referred to as SIP trunk by CUCM) connection to


the IP-PBX, where the call is sent to the appropriate IP-PBX phone. The following steps describe how the outbound call is established as shown in figure 3.

1. A Lync 2010 user initiates a call to the IP-PBX phone.2. The Front End pool sends a SIP INVITE to the Mediation Server.3. The Mediation Server routes the call to CUCM by forwarding the SIP INVITE to CUCM.4. CUCM acknowledges the request.5. CUCM routes the call to the IP-PBX phone.

Figure 3. Outbound Call from Communicator to IP-PBX Phone

When the call is established, the Mediation Server and CUCM will exchange the Session Description Protocol (SDP) to negotiate media capabilities and establish a media path. After the call is answered, media begins to flow.

Note. Figure 3 shows CUCM acting as the Media Termination Point (MTP) resource for the SIP trunk. It’s possible by using media bypass for the media to flow directly between Lync 2010 and CUCM. This is covered later in this chapter.


PBX User Joining Lync Server Audio ConferenceLync Server can create audio conferences that users can join from Lync 2010, a public switched telephone network (PSTN) gateway phone, or an IP-PBX phone. The following steps describe this process as shown in figure 4.

1. An IP phone initiates a call to CUCM to join a Lync Server audio conference.2. CUCM sends a SIP INVITE to the Mediation Server.3. The Mediation Server acknowledges the SIP INVITE by sending a SIP TRYING response

to CUCM.4. The Mediation Server sends a SIP INVITE to the Front End pool where the audio

conference is hosted.5. The Front End pool routes the call to the A/V Conferencing service.

Figure 4. Call from an IP-PBX phone to Lync Server conference bridge


Direct SIP with Media BypassIn Microsoft Office Communications Server, the Mediation Server would remain in the network path of a call for signaling and media traffic when routing the call to an IP-PBX or PSTN gateway. Lync Server removes this restriction. When establishing a call and for its duration, the signaling continues to be routed to the Mediation Server; however, the media no longer has to flow through the Mediation Server. The media traffic can take a more optimum network path. This feature is called media bypass.

There are some technical factors to consider before trying media bypass over a direct SIP connection. Because the media stream flows from a Lync 2010 endpoint to an endpoint that is not a Lync 2010 endpoint, it’s important to note that the following criteria needs to be met to make this a successful scenario:

The IP-PBX must support forked responses to an INVITE. The IP-PBX must support Early-Offer. Media bypass is not supported with Delayed-

Offer (INVITE without SDP).

There are some use cases where media bypass is very helpful in both reducing network utilization and optimizing media paths. An example of this is shown in figure 5. Here, the Lync 2010 user calls the IP-PBX phone that is connected to the CUCM cluster. Using media bypass, the media traffic flows directly between the Lync 2010 client and CUCM before reaching the IP-PBX phone. As a result, the media doesn’t need to traverse the WAN. A more optimum (local) media path is established directly with the IP-PBX (that is, CUCM).


Figure 5. Direct SIP using media bypass

Direct SIP with Simultaneous RingingDirect SIP with simultaneous ringing in Lync Server provides users the ability to receive inbound calls on both Lync 2010 as well as on their IP-PBX phone. During the transition to Lync Server, there may be a period of time when users want to use their enterprise Direct Inward Dialing (DID) number for Lync 2010 and their PBX handset. This allows users to answer the call from their preferred endpoint.

When an inbound call is routed to Lync Server, it sends the SIP INVITE to all the registered endpoints of the callee based on the user’s preferences, including the IP-PBX phone specified by the user. To ring the user’s IP-PBX phone, Lync Server forwards the SIP INVITE


to the IP-PBX by using the Mediation Server across the direct SIP connection as shown in figure 6. It illustrates CUCM as the IP-PBX.

Figure 6. Direct SIP with simultaneous ringing on IP-PBX

Note. The call is being answered on the IP-PBX phone while the media flows from the PSTN gateway, to the Mediation Server, and then to CUCM.


Direct SIP ConfigurationSuccessful direct SIP configuration requires the following:

Configuring the IP-PBX to send and receive SIP signaling and media traffic to and from the Mediation Server.

Configuring the Mediation Server to send and receive signaling (SIP) and media (real-time system protocol (SRTP)) traffic to and from the IP-PBX.

IP-PBX products that are qualified to interoperate with Lync Server for direct SIP are listed in the Microsoft Unified Communications Open Interoperability Program at http://go.microsoft.com/fwlink/?LinkID=187781.

This section covers direct SIP configuration of Lync Server with CUCM as one of the qualified IP-PBX.

Before configuring direct SIP, it is very important to have a dial plan structure defined between Lync Server and the IP-PBX. Typically, the IP-PBX supports a shortened dial plan, with extensions that are often a subset of the user’s DID. There may also be variable length extensions and internal numbers that need to be routed to and from the IP-PBX.

In Lync Server, best practice is to use a dial plan that is based on the E.164 standard. This allows easier routing and troubleshooting as well as a scalable model for growth. It is best practices to represent numbers in an E.164 format (External DIDs as well as internal extensions, which are both covered later in this chapter).

We will look at the individual configurations of Lync Server and CUCM trunk set-up, dial plan considerations, and call routing.

Lync Server ConfigurationLync Server configuration can be done in a couple of ways—through the Microsoft Lync Server Control Panel or through Windows PowerShell®. For the purposes of configuring a direct SIP connection with CUCM, we will illustrate the configuration by using the Lync Server Control Panel and Topology Builder.

Before any configuration changes can be made, the following are required:

CUCM must be added in Topology Builder as a PSTN gateway. PSTN gateway is the generic term to refer to IP-PBX and other gateway devices.

The topology must be published to Central Management store.

After the PSTN gateway (that is, CUCM) is added to the topology, it appears in the Lync Server Control Panel. The first thing to do is to add a trunk to the IP-PBX (that is, CUCM) as shown in figure 7.


Figure 7. Create a trunk to the IP-PBX

Select Pool trunk, and then select the CUCM from the list that appears

On the Trunk Configurations tabbed page, configure the following parameters:

Encryption Level: Refers to whether media encryption should be required, optional, or not supported on the trunk. This setting depends on whether the CUCM is configured for SRTP.

Media Bypass: Enable it if you want to support media bypass for media traffic on this trunk. If media bypass is enabled on this trunk, there are several other settings that must be configured for this to work:

o Enable refer support must be turned off. CUCM neither supports nor handles REFER correctly. REFER cannot be enabled with media bypass.

o The RTCPCallsonHold and RTCPActiveCalls parameters must be turned off. Real-time transport control protocol (RTCP) is a control channel that uses the same RTP channel to monitor the network-specific conditions of the real-time transport protocol (RTP) channel. CUCM doesn’t support RTCP. If these variables are not disabled, the Mediation Server expects keep alives for the RTCP channel from CUCM, which may cause unexpected behavior. The RTCP parameters are not displayed through Lync Server Control Panel and have to be configured by using Windows PowerShell: Set-CsTrunkConfiguration -Identity site:{site name} -EnableBypass $True -RTCPActiveCalls $False –RTCPCallsOnHold $False –EnableSessionTimer $True

o The SessionTimer parameter must be enabled. Because the RTCP channel is disabled, session timers must be enabled so that calls don’t stay up indefinitely in case the call doesn’t get properly torn down.


o Global Configuration for Media Bypass must be configured to Always Bypass under Network Management in Lync Server. This setting cannot be used in conjunction with call admission control (CAC). (See figure 9.)

You can also configure translation rules and number manipulations for the trunk, enabling you to change called-party dialed number identification server (DNIS) information going out the trunk. This is useful if the terminating device of the trunk is accepting numbers in a particular format.

Figure 8 shows the trunk configuration from Lync Server to CUCM.

Figure 8. Trunk configuration from Lync Server to CUCM


Figure 9 Global configuration for media bypass

Note. Enabling Call Admission Control is optional and does not affect the operation of Media Bypass. It is recommended as a best practice to enable Call Admission Control to prevent media over-subscribing of WAN links.

After the trunk is configured, routes can be configured to route PBX extensions to the trunk to CUCM. If PSTN connectivity is configured through CUCM, calls to PSTN from Lync 2010 can be routed to CUCM through the trunk.

For example, the phone number range for Contoso is from +14085550XXX through +14085559XXX. These numbers must be routed to CUCM by creating a new voice route. This configuration is done using a regular expression as shown in figure 10.


Figure 10. Create a route for an IP-PBX trunk


Cisco Unified Communications Manager ConfigurationTo configure a SIP trunk in CUCM involves the following procedures. They are configured in the CUCM Admin interface:

1. Create a SIP trunk.2. Specify the Media Resource Group List.3. Configure outbound parameters for the trunk.4. Change significant digits and Calling Search Space.5. Set SIP trunk security profile information.

A SIP trunk must be added to CUCM that routes calls to the Lync Mediation Server (that is, a pool if the Mediation Server role is collocated).

Create a SIP trunk1. Click Device, and then click Trunk. Set Trunk Type to SIP Trunk and set Device

Protocol to SIP as shown in figure 11. 2. Click Next to continue the configuration process.

Figure 11. Create a SIP trunk on the IP-PBX

Specify the Media Resource Group ListThe Media Resource Group list contains a group of media resources, should any communication on the trunk require them to be used. This includes but is not limited to MTP resources, transcoders, conference bridges, Annunciators, Music on Hold resources, and so on. The Media Resource Group List should be a list of transcoders and MTP resources you intend to use for the SIP trunk between Lync Server and CUCM.

Check Media Termination Point Required to enable this setting. Ensure that enough MTP resources are available for the interoperability (see figure 12).


Figure 12. Specify the Media Resource Group List

Enabling the Media Termination Point Required check box has a couple of considerations to be aware of. MTP resources can be required in some scenarios; for example, DTMF relay methods cannot be negotiated or a codec selection doesn’t match on the endpoints. Beginning with CUCM version 5.x, MTP resources can be dynamically allocated if any of the previous circumstances occur.

CUCM, by default, does only Delayed-Offer for outbound calls on a SIP trunk. To do early media on outbound calls (that is, Early-Offer), Media Termination Point Required must be enabled. Although Early-Offer is not required by the Mediation Server, Early-Offer is required for media bypass and enabling Early-Offer is best practice because there are call flows that may be affected by delayed media. Additionally, mid-call updates may also be adversely affected (dropped or unable to connect) for some call flows, especially those that invoke supplementary features.


Configure parameters for a SIP trunkIn the Inbound Calls sections of the Trunk Configuration page, configure the Calling Search Space so the trunk has a set of partitions to search through for the dialed number as shown in figure 13.

Figure 13. Configure inbound parameters for SIP trunk

Change significant digits and Calling Search SpaceCalls routed from the Mediation Server to CUCM can be manipulated by altering significant digits, as well as changing the Calling Search Space for those calls to match a certain set of translation patterns, route patterns, or directory numbers. (This is covered in more detail in the Dial Plan section of this chapter.)

In the SIP Information section of the Trunk Configuration page, set the following as shown in figure 14:

Destination Address—specify the IP address or FQDN of the Mediation Server. This FQDN can be an SRV address, which points to a pool where the Mediation service is collocated.

SIP Trunk Security Profile—configuring a security profile is explained in the following section.


Figure 14. Set Destination Address and SIP Trunk Security Profile

Set SIP Trunk Security Profile InformationIn the SIP Trunk Security Profile Information section, specify the following settings as shown in figure 15:

Set Device Security Mode to Encrypted. This ensures that CUCM uses TLS to communicate with the Mediation Server.

Set Incoming Transport Type to TLS. Set Outgoing Transport Type to TLS. Set Incoming Port to 5061.


Figure 15. Configure Security Profile for SIP trunk to Lync

The CUCM SIP trunk configuration is now complete. The dial plan can be defined to route inbound and outbound calls between the two systems.

Dial PlanDial plan is an important part of creating interoperability between two telephony systems. Each system’s dial plan must not conflict with the other system’s dial plan, and must adhere to inbound and outbound requirements of the other system. A direct SIP implementation has the same dial plan considerations.

This section covers an example of the Contoso dial plan and outlines what is required to implement this dial plan in each system. Figure 16 represents the Contoso dial plan.


Figure 16. Dial plan representation for Contoso

Outbound from Lync ServerLync Server normalizes all outbound calls as E.164. This allows uniform routing that scales globally across the Lync Server deployment. Phone numbers are normalized to E.164 by normalization rules.

In the Contoso example, the company has standardized on five digits for the length of their extensions. When a Lync Server user dials an IP-PBX phone, they dial the five-digit extension that is assigned to the IP-PBX phone. Lync 2010 normalizes this extension into the E.164 format by using the normalization rules that are defined in the caller’s (that is, the Lync 2010 user) location profile. This is shown in figure 17, where extension 50135 is normalized to +1-425-555-0135.

Figure 17. Normalization of an outbound extension


This E.164 phone number is then matched to a voice route defined in Lync Server by the administrator. This voice route is configured to route the call to CUCM.

The dialed number matches the pattern that is defined in the voice route, and then Mediation Server sends a SIP INVITE to CUCM (defined as HQ-PstnGateway-1) through the Mediation Server with the TO: header set to the called number, +14255550135.

After CUCM receives the call, it needs to route the call to the IP-PBX endpoint. Because Contoso is using Cisco Unified Communications Manager 7.x, CUCM is E.164-compliant and therefore can accept the plus sign as a character in a called/calling party number. The Mediation Server doesn’t need to strip the plus sign from the number before sending it to CUCM. However, Cisco Unified Communications Manager versions prior to 7.x aren’t E.164-compliant and are unable to route phone numbers preceded by a plus sign. For versions of CUCM earlier than 7.x, there are two options for removing the plus sign before routing the call to CUCM.

Option 1The plus sign can be stripped off by the Mediation Server before the call is sent to CUCM. It is then able to route the call to the destination IP-PBX phone. This is the most straightforward method to route calls to Cisco Unified Communications Manager versions prior to 7.x.

To manipulate the called party number before routing it to CUCM, specify one or more translation rules on the trunk configuration as shown in figure 18.


Figure 18. Translation Rule on a trunk for removing the plus sign

Alternatively, you can also use the Windows PowerShell parameter RemovePlusfromURIby using Set-CsTrunk. That removes the plus from the automatic number identification (ANI) and DNIS before sending it out on the trunk.

Option 2You can configure the CUCM trunk to consider only the last 11 digits of the called number. This is done by setting the Significant Digits parameter in the Inbound Calls section of the Trunk Configuration as shown in figure 19.

Figure 19. Set significant digits on CUCM

This configuration option is not recommended because it’s a static configuration on the trunk and doesn’t work if you allow variable length phone numbers (that is, extension numbers, international numbers, and non-U.S. numbers) across the trunk.

CUCM also needs to be configured to route the call after it receives it on the trunk. CUCM uses the concept of Calling Search Spaces and Partitions. It enforces which directory


numbers, route patterns, or translation patterns can be reached by the trunk. This is important because it puts the trunk in its own dialing domain.

A Calling Search Space should be created that contains a partition with various translation rules. They are responsible for converting the number to a format that can be routed by CUCM, whether the number is to an endpoint extension or to PSTN. This is similar to the concept of normalization on Lync Server. This is shown in figure 20.

Figure 20. Calling Search Space configuration on CUCM


Figure 21 shows the transaction rule configuration.

Figure 21. Translation rule configuration

In the Contoso example, the plus sign is used in the translation pattern because a full E.164 number is sent to CUCM for processing. Translation rules also have a Calling Search Space (calling domain). Each translation rule, after it translates the number to the appropriate format, contains a Calling Search Space that contains the phone’s partition for routing to the phone endpoints or partitions that contain route patterns that route to the appropriate PSTN gateways.


Inbound to Lync ServerWhen Lync Server receives a call from CUCM, it could be in multiple formats, including five-digit extensions. The number must be normalized to E.164 format so it can be routed appropriately to a Lync 2010 user, a Conferencing Attendant, Exchange Auto Attendant, or a Response Group. This is done by configuring a pool dial plan to the Mediation Server with the appropriate normalization rules to convert the inbound called party numbers to E.164. Figure 22 shows the pool dial plan on the Mediation Server.

Figure 22. Pool dial plan on the Mediation Server


SummaryDirect SIP is the most straightforward way to achieve interoperability between an IP-PBX and Lync Server. Direct SIP provides a standards-based interface for signaling and media for audio communications. Direct SIP supports the following use cases:

Use Simultaneous Ring feature on IP-PBX or Lync Server to ring both IP-PBX handset and Lync 2010

Call from IP-PBX handset to Lync 2010 Call from Lync 2010 to IP-PBX handset Call from IP-PBX handset to dial-in conferencing on Lync Server Call from IP-PBX handset to Lync Server Response Group PSTN access for Lync Server through IP-PBX


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