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    Avaya Solution & Interoperability Test Lab

    Sample Configuration Illustrating Avaya Aura

    Communication Manager SIP Trunking Using Processor

    Ethernet and Acme Packet Net-Net 4500 Session Director

    Issue 1.0

    Abstract

    These Application Notes illustrate a sample configuration of Avaya Aura Communication

    Manager Release 5.2 with SIP Trunks to Acme Packet Net-Net 4500 Session Director at twosites. For business continuity, a primary site uses Avaya S8730 Servers, and a secondary site

    uses an Avaya S8500 Server as an Enterprise Survivable Server (ESS). At each site, two

    Avaya C-LAN cards as well as the Processor Ethernet of the sites Avaya Server are

    configured for SIP Trunking to the inside realm of an Acme Packet Net-Net 4500 SessionDirector. On the outside realm, each Acme Packet Net-Net 4500 Session Director is

    connected to a SIP network simulating a public SIP Service Provider. Within each site, theAcme Packet Net-Net 4500 Session Director is configured for load spreading and fast fail-over

    of inbound calls to the enterprise from the PSTN. For outbound calls to the PSTN,

    Communication Manager is configured for location-based routing for trunk selection andLook-Ahead Routing for trunk fail-over.

    These Application Notes complement previously published Application Notes, focusing on thenew Communication Manager 5.2 capability to use the Processor Ethernet of the Avaya

    S8730 Servers and Avaya ESS as the Avaya interface for SIP Trunk signaling.

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    1. IntroductionThese Application Notes illustrate a sample configuration of Avaya Aura Communication

    Manager Release 5.2 with SIP Trunks to Acme Packet Net-Net 4500 Session Director at two

    sites. A primary site uses Avaya S8730 Servers, and a secondary site uses an Avaya S8500

    Server as an Enterprise Survivable Server (ESS). At each site, two Avaya C-LAN cards as wellas the Processor Ethernet of the sites Avaya server are configured for SIP Trunking to the

    Session Director using TCP for the SIP signaling connectivity. Each Session Director is also

    connected to a SIP network simulating a SIP Service Provider. Since most SIP Service Providersuse UDP for SIP signaling, the SIP signaling connectivity from the Acme Packet Net-Net 4500toward the outside realm uses UDP.

    These Application Notes complement previously published Application Notes [JRR] that

    covered the C-LAN based SIP Trunking and survivability considerations when various failuresare induced. Figure 1is repeated from reference [JRR], but the detailed configuration and call

    verifications from reference [JRR] will not be repeated in these Application Notes. These

    Application Notes build upon the configuration documented in [JRR] by adding SIP Trunkingfrom the Processor Ethernet of the Avaya S8730 Servers, as well as the Avaya S8500 ESS.

    Figure 1: Avaya Aura Communication Manager Survivable SIP Trunking

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    The network shown in Figure 1remains in place for the verification of these Application Notes.

    Figure 2omits many of the details, but illustrates (in orange) the addition of two new SIP trunks(32, 62) via the Processor Ethernet of the Avaya S8730 Servers and Avaya S8500 ESS.

    Figure 2: Avaya Aura Communication Manager Processor Ethernet SIP Trunking

    Although a mix of C-LAN based SIP trunks and Processor Ethernet based SIP trunks are used inthe sample configuration, the use of the Avaya S8730 Server Processor Ethernet does not require

    Avaya G650 Media Gateways. In some configurations, Processor Ethernet can obviate the need

    for Avaya G650 Media Gateways and the associated IPSI, C-LAN, and Media Processor cards.Although not the subject of these Application Notes, Avaya H.248 gateways (e.g., Avaya G250,

    G350, G450, or G430 Media Gateways) and Avaya H.323 Telephones can register directly to the

    Avaya S8730 Processor Ethernet, beginning in Communication Manager 5.2. A mix of C-LAN

    based SIP trunks and Processor Ethernet based SIP trunks are used in the sample configuration,since it is expected that large customers interested in deploying a multi-site survivable SIPTrunking solution may either already have Avaya G650 Media Gateways or require Avaya G650

    Media Gateways for scale or specific application reasons. Combining the more familiar C-LAN

    based SIP trunks with new Processor Ethernet based SIP trunks in the same configuration also

    provides an opportunity to compare and contrast behaviors for instructional purposes.

    The Acme Packet Net-Net 4500 is used to distribute SIP signaling for incoming calls to multipleC-LAN interfaces and the Processor Ethernet of the sites Avaya server, providing loadspreading and fast automatic fail-over. The Acme Packet Net-Net 4500 performs conversion

    between TCP transport for SIP signaling used by Communication Manager to UDP transport

    commonly used by SIP Service Providers. The Acme Packet Net-Net 4500 also performsSession Border Controller (SBC) functions, providing security and topology-hiding at the

    enterprise edge. In the sample configuration, all SIP signaling and RTP media between the

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    enterprise and the (simulated) SIP Service Provider flows through the Acme Packet Net-Net

    4500.

    A customer interested in SIP Trunk survivability may want a redundant pair of Acme Packet

    Net-Net 4500 Session Directors at each site. Although the sample configuration verified in these

    Application Notes used only a single Acme Packet Net-Net 4500 at each site, the Acme Packetconfiguration shown in Appendix A of reference [JRR] was prepared as if there were a high

    availability Acme Packet configuration at each site. Actual verification testing of the Acme

    Packet Net-Net 4500 High Availability configuration with Communication Manager wasperformed as part of Avaya DevConnect compliance testing, and the Application Notes in

    reference [AP-HA] documents the configuration and testing results.

    1.1. Summary of Inbound Calls to the Enterprise

    Figure 2is repeated from reference [JRR] to illustrate aspects of the sample configurationrelated to inbound calls to the enterprise from the PSTN. Although further elaboration of the

    simulation of the SIP Service Provider is out of scope, Figure 2 may help with understanding

    assumptions and the call flow verifications in Section 5. For example, it is assumed thatpublished PSTN telephone numbers such as Direct Inward Dial (DID), Listed DirectoryNumbers (LDN), or toll-free numbers that map to Avaya Vector Directory Numbers (VDNs) can

    arrive to the enterprise from the service provider via either the primary site or the secondary site

    SIP Trunks. A SIP Service Provider may load balance inbound calls to the enterprise betweensites, or route particular numbers to a specific site preferentially, with fail-over to the other site

    as needed.

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    As shown in Figure 3, the DID number 732-852-1816 is preferentially routed to the primary site,

    but can fail-over to the secondary site. The DID number 732-852-2940 is preferentially routed tothe secondary site, but can fail-over to the primary site. Communication Manager can map any

    received telephone number to any destination via the incoming call handling table of the trunk

    group receiving the call. During testing, calls arriving via the primary site SIP trunks were

    directed to users at both the primary and the secondary sites, and vice-versa.

    Figure 3: Incoming Calls to the Enterprise from the PSTN

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    1.2. Session Director and Communication Manager Terminology

    The table below is repeated from reference [JRR] to provide a translation for key concepts and

    terminology that may be helpful to readers familiar with Acme Packet Session Director or Avaya

    Aura Communication Manager, but not both. Of course, these analogies are imperfect, so the

    table is intended only as a starting point in understanding.

    Acme Packet

    Concept

    Avaya Aura

    Communication

    Manager Concept

    Notes

    Session-agent(SIP)

    SIP Signaling Group andSIP Trunk Group

    The session-agent defines a SIP peer, similar toan Avaya SIP signaling group/trunk group.

    Session-agent

    group (SAG)

    Route-Pattern with

    multiple SIP Trunk

    Groups

    Session agents can be configured in a SAG, so

    that routing to the SAG can use any session

    agent in the group. The analogy is imperfect inthat the SAG can use strategies that select agents

    from the SAG based on real-time trafficconditions and defined constraints (i.e., more

    sophisticated outbound load balancing options)

    Sag-recursion Look-Ahead Routing(LAR) for a route-

    pattern

    If a session agent is selected, but adownstream failure results, sag-recursion can

    trigger Session Director to automatically use a

    different session agent in the SAG. This issimilar to Communication Manager selection of

    a SIP trunk group, followed by a downstream

    signaling failure causing LAR (route-advance)to the next trunk in the route pattern.

    Session-agentconfiguration for

    ping-method

    OPTIONS

    Enable Layer 3 Test =y for the SIP Signaling

    Group

    Both Communication Manager and AcmePacket can be configured to source OPTIONS

    messages to check the health of a peer. See

    Section 1.3 for more information.

    To summarize the Acme Packet Net-Net 4500 configuration, a session agent is defined for

    each C-LAN at the site, and the processor Ethernet of the Avaya S8730 Server or AvayaS8500 ESS, depending on the site. A session agent group (SAG) is configured to group the

    session agents, and a strategy for distribution of calls to the session agents that are members of

    the group is specified. In the sample configuration, once an inbound call reaches an enterprise

    site, the Session Director is configured for round-robin call distribution to the members of the

    session agent group. If connectivity to a particular session agent fails, and the Acme Packet Net-Net 4500 has not yet detected the failure, an inbound call directed to the failed session agent will

    encounter a transaction timeout. The transaction timeout will cause the call to be directed toanother session agent at the same site automatically. In addition, the failed session agent will be

    marked out-of-service so that subsequent inbound calls will flow to an operational session agent

    without experiencing the timeout. If all session agents that are part of the session agent groupexperience a transaction timeout, then a SIP 408 message would be returned to the SIP Service

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    Provider. If all session agents that are part of the session agent group are already marked out of

    service, then a SIP 503 Service Unavailable would be returned to the SIP Service Provider.Similarly, if the public side of the Session Director experienced failures for an outbound call

    from Communication Manager, Communication Manager would receive no SIP response after

    100 TRYING, a 408 Transaction Timeout, or a 503 Service Unavailable, depending on the

    particular failure scenario. Note that all these conditions are triggers for CommunicationManager Look-Ahead Routing. Reference [LAR] documents another sample configuration for

    Look-Ahead Routing, and includes a more complete list of SIP triggers.

    In the sample configuration, the Acme Packet Net-Net 4500 at a given site does not have session

    agents to the other site. It is presumed that a production SIP Service Provider can redirect calls

    from one site to another based on failure conditions, such as a timeout, or the return of a 503Service Unavailable.

    1.3. SIP OPTIONS Message, Service States, and Call Acceptance

    Both Communication Manager and the Acme Packet Net-Net 4500 can use a SIP OPTIONS

    message to verify connectivity health. This section summarizes the use of the SIP OPTIONSmessage, the implications for in-service and out-of-service determinations, and the effect on newcall attempts. See Section 5.7 of reference [JRR] as well as Section 5.7 of these Application

    Notes for Wireshark traces related to the topics in this section.

    In the sample configuration, the Acme Packet Net-Net 4500 is configured to periodically check

    the availability of a session agent via a SIP OPTIONS message. The interval between SIP

    OPTIONS messages is configurable. By default, any SIP response would be considered anacceptable reply, including normal 200 OK responses, but also other responses such as 503

    Unavailable. The responses from Communication Manager deemed acceptable for marking the

    session agent in-service can be configured, if desired. Although a failed SIP OPTIONS

    exchange can result in a session agent being marked out-of-service, in a system with continuouscall activity, it would be more likely that a transaction timeout for a SIP method such as INVITE

    would cause a recently failed session agent to be marked out-of-service. In this light, the SIP

    OPTIONS exchange is more likely to be the method of bringing a previously failed session-agentback in service. Therefore, if rapid recovery from prior failures is paramount, the time between

    SIP OPTIONS generated by the Acme Packet Net-Net 4500 can be reduced to a low value. In

    the sample configuration, testing was done with a 60 second interval, and later with a 16 secondinterval.

    If the Acme Packet Net-Net 4500 has marked a session-agent out-of-service, the session agentwill not be chosen for call activity. That is, if both an out-of-service session agent and an in-

    service session agent appear in the same session agent group, the in-service session agent willnaturally be chosen for the next call. The out-of-service session agent can come back in-service

    either via a SIP response to an Acme Packet sourced SIP OPTIONS message, or due to a SIPmessage, such as an INVITE or OPTIONS received from the session agent. Indeed, if an

    INVITE message for a Communication Manager outbound call is received by the Acme Packet

    Net-Net 4500 from a session agent that had been marked out-of-service, the INVITE isprocessed, the call can succeed, and the session agent is again marked in-service. For the reader

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    familiar with Avaya trunk states, this is similar to the Communication Manager behavior for a

    trunk marked in the Out-of-service/Far-end state.

    When an Avaya SIP signaling group is marked with Enable Layer 3 Test = y,

    Communication Manager will periodically send a SIP OPTIONS method to the far-end of the

    signaling group. When the Acme Packet Net-Net 4500 receives such a SIP OPTIONS, it checksthe logical next-hop. In the sample configuration, the next hop is the SIP Service Provider.

    If there is no in-service next-hop, then the Acme Packet Net-Net 4500 returns a 503 Service

    Unavailable to Communication Manager. Communication Manager will then mark the SIPsignaling group for bypass, and the corresponding SIP trunk group will be marked Out-of-

    service/Far-end. For example, if the Acme Packet Net-Net 4500 has detected that the SIP

    Service Provider network has failed, then a SIP OPTIONS from the Avaya session agent willreceive a 503, and the Avaya trunks will be marked for bypass, which is appropriate. In this

    state, although outbound calls from the enterprise will not select the trunk, if an inbound call is

    received, the network has apparently recovered. The call will be accepted, and the Avaya SIP

    trunk group will be marked in-service.

    1.4. Summary of Outbound Calls from the Enterprise to the PSTN

    For outbound calls, Communication Manager location-based routing can direct outgoing calls

    from users at a given site to the SIP trunks in the same site, with fail-over to use SIP trunks at the

    other site as needed. The user dials the Automatic Route Selection (ARS) access code followedby the PSTN number. In the sample configuration, if a user at the primary site dials the number,

    the call will be directed to route-pattern 30, which lists the SIP trunks at the primary site first. If

    the trunks at the primary site are unable to take the call, either due to congestion or failure, thecall will proceed out the trunks at the secondary site, which are also members of route-pattern 30.

    Outbound calls placed from users at the secondary site are directed to route-pattern 60, which

    lists the SIP trunks at the secondary site first, with overflow and fail-over to the SIP trunks at the

    primary site.

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    2. Equipment and Software ValidatedThe following equipment and software were used for the sample configuration provided:

    Equipment Software

    Avaya S8730 Servers (Main)

    Avaya Aura Communication Manager

    Release 5.2 (947.3) with and without SP1

    (patch 17294)

    Avaya S8500 Server (ESS)

    Avaya Aura Communication Manager

    Release 5.2 (947.3) with and without SP1(patch 17294)

    Avaya 4600-Series Telephones (H.323) Release 2.9 H.323

    Avaya 9600-Series Telephones (H.323) Release 3.0 H.323

    Avaya 2400-Series Digital Telephones N/A

    Acme Packet Net-Net 4500 Session Directors CX6.1.0 patch 3

    Cisco AS5400 Universal Gateway 12.4(15)T7

    Cisco 3825 Integrated Services Router 12.4(11)XW7

    Table 1: Equipment and Software Versions Used

    3. Avaya Aura Communication Manager ConfigurationThe network shown in Figure 1 is documented in reference [JRR]. Rather than repeat the

    configuration, this section focuses on the changes to the configuration to allow SIP signaling via

    the Processor Ethernet of the Avaya S8730 Servers and Avaya S8500 ESS. Productdocumentation can be found in references [CM1], [CM2], [CM3], and [ESS]. Except for the

    web page configuration shown in Section 3.1, all remaining configuration is performed via theCommunication Manager SAT interface. Screens are abridged for brevity in presentation.

    3.1. Conf iguration Processor Ethernet Via the Web Pages

    The Processor Ethernet must be configured via the Configure Server Web pages on both

    S8730 servers. The screens in this section illustrate the configuration on only one of the servers.

    Consult product documentation for further procedural guidance (e.g., performing changes on thestandby server to mitigate service disruptions).

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    The S8730 Server can be accessed via a web interface in an internet browser. The following

    screen shows the initial screen. To add Processor Ethernet to an existing configuration such asthe configuration documented in [JRR], select Configure Server under Installation, as shown.

    Navigate to Set Identities. The following screen shows the lower portion of the Set Identities

    page before any changes are made. Note that the Processor Ethernet (PE)drop-down initiallyindicates UNUSED.

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    The following screen shows the lower portion of this same screen, where Ethernet 2 is being

    selected in the Processor Ethernet (PE)drop-down. In this case, Ethernet 2 is the CorporateLAN interface, and the Processor Ethernet and Corporate LAN interface will be the same.

    Select the Continuebutton to proceed. The screen below shows a portion of the resulting

    screen, with the IP Addresses from the sample configuration populated in the mandatory fields,

    indicated by a red star. The active server IP Address is 2.2.87.13. This is the IP Address thatwill be associated with the Processor Ethernet or procr of the active S8730 Server.

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    At the conclusion of the process, a screen such as the following should be displayed.

    At the conclusion of the process on both S8730 Servers, the Status Summarypage may be usedto check the Processor Ethernet status, as shown below.

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    3.2. Node Names

    Node names are mappings of names to IP Addresses that can be used in various screens. The

    following abridged list output shows some of the relevant node-names in the sample

    configuration. The entry for the Processor Ethernet called procr is shown in bold. Note that

    the procr node-name appears automatically due to the web configuration in Section 3.1.

    list node-names all Page 1NODE NAMES

    Type Name I P Addr essI P nn4500- pr i si t e 2. 2. 85. 45I P nn4500- secsi t e 2. 2. 185. 145IP procr 2.2.87.13

    I P t n2602- 1a11 2. 2. 185. 4I P tn2602- 2a07 2. 2. 26. 3I P t n2602- 2b07 2. 2. 26. 2I P val - 1a07 2. 2. 185. 25I P val - 2a08 2. 2. 85. 25

    3.3. Adding the IP Interface for procr

    The add ip-interface procr command is used to configure the Processor Ethernet parameters.

    The following screen shows the parameters used in the sample configuration. Although not the

    focus of these Application Notes, it can be observed that the Processor Ethernet may also be usedfor registrations from H.323 IP Telephones and H.248 gateways.

    add ip-interface procr Page 1 of 1I P I NTERFACES

    Type: PROCRTarget socket l oad: 19200

    Enable Interface? y Al l ow H. 323 Endpoi nts? yAl l ow H. 248 Gat eways? y

    Network Region: 3 Gat ekeeper Pr i or i t y: 5

    I PV4 PARAMETERSNode Name: procr

    Subnet Mask: / 24

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    The following screen lists the IP interfaces, which now includes the bold procr at the end of

    the list.

    list ip-interface all Page 1I P I NTERFACES

    NetON Type Sl ot Code/ Sf x Node Name/ Mask Gat eway Node Rgn VLAN

    I P- Addr ess- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -y C- LAN 01A02 TN799 D c- l an / 24 Gat eway001 1 n

    2. 2. 185. 2y C- LAN 02A02 TN799 D c- l an2a02 / 24 Gat eway002 3 n

    2. 2. 85. 2y VAL 01A07 TN2501 val - 1a07 / 24 Gat eway001 n

    2. 2. 185. 25y MEDPRO 02A07 TN2602 t n2602- 2a07 / 24 Gat eway003 3 n

    2. 2. 26. 3y MEDPRO 02B07 TN2602 t n2602- 2b07 / 24 Gat eway003 3 n

    2. 2. 26. 2y MEDPRO 01A11 TN2602 t n2602- 1a11 / 24 Gat eway001 1 n

    2. 2. 185. 4y C- LAN 02B02 TN799 D c- l an2b02 / 24 Gat eway002 3 n

    2. 2. 85. 20y C- LAN 01A10 TN799 D c- l an1A10 / 24 Gat eway001 1 n

    2. 2. 185. 20y VAL 02A08 TN2501 val - 2a08 / 24 Gat eway002 n

    2. 2. 85. 25y PROCR 2.2.87.13 /24 2.2.87.1 3

    3.4. SIP Signaling Groups Using Processor Ethernet

    This section illustrates the configuration of the SIP Signaling Groups that use the Processor

    Ethernet. The SIP Signaling Groups that use a C-LAN are documented in reference [JRR]. Twoadditional signaling groups are defined. Each signaling group has a Group Type of sip, and

    a Near-end Node Name of procr. The Transport Method for all signaling groups is tcp

    using port 5060. The Far-end Domain for each signaling group is the inside IP Address ofthe appropriate Acme Packet Net-Net 4500. The Enable Layer 3 Test field is enabled to allowCommunication Manager to maintain the signaling group using the SIP OPTIONS method, as

    described in Section 1.3. Other fields can be left at default values, including DTMF over IP

    set to rtp-payload, which corresponds to RFC 2833. Note that the Alternate Route Timerthat defaults to 6 seconds impacts fail-over timing for outbound calls. If Communication

    Manager does not get an expected response, Look-Ahead Routing can be triggered, after the

    expiration of the Alternate Route Timer.

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    The following screen shows the addition of signaling group 32. The Near-end Node Name is

    procr and the Far-end Node Name is the node name of the Acme Packet Net-Net 4500 at theprimary site. The Far-end Network Region is configured to region 3. In normal operation,

    when all network elements are up, signaling group 32 will be in-service, under the control of the

    active S8730 Server. If the ESS becomes active, signaling group 32 will be out-of-service from

    the point of view of the ESS. The ESS will never receive a response from the primary siteSession Director at the far-end of signaling group 32.

    add signaling-group 32 Page 1 of 1

    Gr oup Number : 32 Group Type: sipTransport Method: tcp

    I MS Enabl ed? n

    Near-end Node Name: procr Far-end Node Name: nn4500-prisite

    Near-end Listen Port: 5060 Far-end Listen Port: 5060

    Far-end Network Region: 3

    Far-end Domain: 2.2.85.45

    Bypass I f I P Thr eshol d Exceeded? n

    DTMF over I P: r t p- payl oad Di r ect I P- I P Audi o Connect i ons? ySessi on Establ i shment Ti mer( mi n) : 3 I P Audi o Hai r pi nni ng? n

    Enable Layer 3 Test? y Di r ect I P- I P Ear l y Medi a? nH. 323 St at i on Out goi ng Di r ect Medi a? n Al t ernat e Rout e Ti mer( sec) : 6

    The following screen shows the addition of signaling group 62. The Near-end Node Name isprocr and the Far-end Node Name is the node name of the Acme Packet Net-Net 4500 at the

    secondary site. The Far-end Network Region is configured to region 1. In normal operation,

    when all network elements are up, signaling group 62 will be out-of-service from the point of

    view of the active S8730 Server, which will never receive a SIP 200 OK response from thesecondary site Session Director at the far-end of signaling group 62. However, when the ESS

    is active, signaling group 62 will come in-service from the point of view of the ESS.

    add signaling-group 62 Page 1 of 1

    Gr oup Number : 62 Group Type: sipTransport Method: tcp

    I MS Enabl ed? n

    Near-end Node Name: procr Far-end Node Name: nn4500-secsite

    Near-end Listen Port: 5060 Far-end Listen Port: 5060Far-end Network Region: 1

    Far-end Domain: 2.2.185.145

    Bypass I f I P Thr eshol d Exceeded? n

    DTMF over I P: r t p- payl oad Di r ect I P- I P Audi o Connect i ons? y

    Sessi on Establ i shment Ti mer( mi n) : 3 I P Audi o Hai r pi nni ng? nEnable Layer 3 Test? y Di r ect I P- I P Ear l y Medi a? n

    H. 323 St at i on Out goi ng Di r ect Medi a? n Al t ernat e Rout e Ti mer( sec) : 6

    3.5. SIP Trunk Groups

    This section illustrates the configuration of two additional SIP Trunks Groups, corresponding to

    the two SIP signaling groups using Processor Ethernet. Each trunk group has a Group Type of

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    sip. The two new SIP trunk groups are conceptually the same as those defined for trunk

    groups 30, 31, 60, and 61 in reference [JRR].

    The following shows page 1 for trunk group 32. The Number of Members field defines how

    many simultaneous calls are permitted for the trunk group, and can be coordinated with Acme

    Packet Net-Net 4500 call admission control features if desired.

    add trunk-group 32 Page 1 of 21TRUNK GROUP

    Gr oup Number : 32 Group Type: sip CDR Repor t s: yGroup Name: SIP-PSTN-PE COR: 1 TN: 1 TAC: 132Direction: two-way Outgoi ng Di spl ay? n

    Di al Access? n Ni ght Servi ce:Queue Length: 0Service Type: public-ntwrk Aut h Code? n

    Signaling Group: 32

    Number of Members: 10

    The following shows Page 2 for trunk group 32. All parameters shown are default values, exceptfor the Preferred Minimum Session Refresh Interval, which has been changed from 600 to 900

    to avoid unnecessary SIP messaging with the Cisco products used to simulate the SIP Service

    Provider. As such, this screen will not be shown for trunk group 62.

    add trunk-group 32 Page 2 of 21Gr oup Type: si p

    TRUNK PARAMETERSUni code Name: yes

    Redi r ect On OPTI M Fai l ur e: 5000SCCAN? n Di gi t al Loss Gr oup: 18

    Preferred Minimum Session Refresh Interval(sec): 900

    The following shows Page 3 for trunk group 32. All parameters shown are at default values. Assuch, this screen will not be shown for trunk group 62.

    change trunk-group 30 Page 3 of 21TRUNK FEATURES

    ACA Assi gnment? n Measur ed: noneMai ntenance Test s? y

    Number i ng Format : publ i cUUI Treat ment : servi ce- pr ovi der

    Repl ace Rest r i ct ed Number s? nRepl ace Unavai l abl e Number s? n

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    The following shows Page 4 for trunk group 32. All parameters shown are at default values. As

    such, this screen will not be shown for trunk group 62. Depending on the Service Provider, itmay be necessary to enter a specific value, such as 101, in the Telephone Event Payload Type

    associated with DTMF signaling. Check with the specific service provider. Similarly, some

    Service Providers may require that the fields Support Request History and Send Diversion

    Header be changed from default values for proper support of redirection features such asExtension to Cellular or call forwarding off-net.

    change trunk-group 30 Page 4 of 21PROTOCOL VARI ATI ONS

    Mar k User s as Phone? nPrepend ' +' t o Cal l i ng Number ? n

    Send Tr ansf err i ng Part y I nf ormati on? nSend Di versi on Header? n

    Suppor t Request Hi st ory? yTel ephone Event Payl oad Type:

    The following shows Page 1 for trunk group 62. Remaining pages match trunk group 32.

    add trunk-group 62 Page 1 of 21TRUNK GROUP

    Gr oup Number : 62 Group Type: sip CDR Repor t s: yGroup Name: SIP-PSTN-ESS-PE COR: 1 TN: 1 TAC: 162Direction: two-way Outgoi ng Di spl ay? n

    Di al Access? n Ni ght Servi ce:Queue Length: 0Service Type: public-ntwrk Aut h Code? n

    Signaling Group: 62

    Number of Members: 10

    3.6. Route Patterns

    Route pattern 30 will be used for calls that prefer the SIP trunks at the primary site, but may useSIP trunks at the secondary site if the SIP trunks at the primary site are busy or failed. Note thatLook-Ahead Routing (LAR) is set to next. As an example of LAR, assume the Acme Packet

    4500 at the primary site has just failed, and Communication Manager has not yet marked primary

    site trunks out-of-service. Assume that an outbound call is made that chooses this route-pattern.

    The call can use LAR for automatic route-advance to complete successfully using the SIPtrunks at the secondary site. Digit manipulation can be performed on the number, if needed. In

    the sample configuration, the leading digit (i.e., the 1) is deleted and a 10 digit number is sent.

    (This may not be representative of the numbering scheme expected by a production SIP ServiceProvider.) In Section 5, the order of the trunks in the route-pattern is varied for specific

    verifications.

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    change route-pattern 30 Page 1 of 3Patt ern Number : 30 Pattern Name: SIP-PSTN-P

    SCCAN? n Secur e SI P? nGr p FRL NPA Pf x Hop Tol l No. I nser t ed DCS/ I XCNo Mr k Lmt Li st Del Di gi t s QSI G

    Dgts I ntw1: 32 0 1 n user

    2: 31 0 1 n user3: 30 0 1 n user4: 60 0 1 n user5: 61 0 1 n user6: 62 0 1 n user

    BCC VALUE TSC CA- TSC I TC BCI E Servi ce/ Feat ure PARM No. Numberi ng LAR0 1 2 M 4 W Request Dgts Format

    Subaddress1: y y y y y n n r est next2: y y y y y n n r est next3: y y y y y n n r est next4: y y y y y n n r est next5: y y y y y n n r est next6: y y y y y n n r est none

    Route pattern 60 will be used for calls that prefer the SIP trunks at the secondary site, but may

    use the SIP trunks at the primary site, if the SIP trunks at the secondary site are busy or failed.

    As with route-pattern 30, LAR is configured to next to allow calls to complete automatically infailure scenarios. Although not necessary, trunk group 62, associated with the Processor

    Ethernet of the ESS, is listed after the C-LAN based trunks at the secondary site, since trunk

    group 62 will be out-of-service during normal operation.

    change route-pattern 60 Page 1 of 3Patt ern Number : 60 Pattern Name: SIP-PSTN-S

    SCCAN? n Secur e SI P? nGr p FRL NPA Pf x Hop Tol l No. I nser t ed DCS/ I XC

    No Mr k Lmt Li st Del Di gi t s QSI GDgts I ntw

    1: 61 0 1 n user2: 60 0 1 n user3: 62 0 1 n user4: 31 0 1 n user5: 30 0 1 n user6: 32 0 1 n user

    BCC VALUE TSC CA- TSC I TC BCI E Servi ce/ Feat ure PARM No. Numberi ng LAR0 1 2 M 4 W Request Dgts Format

    Subaddress1: y y y y y n n r est next2: y y y y y n n r est next3: y y y y y n n r est next

    4: y y y y y n n r est next5: y y y y y n n r est next6: y y y y y n n r est none

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    3.7. Administer Publ ic Numbering

    The change public-unknown-numbering command may be used to define the format of the

    calling party number to be sent. In the bolded rows shown in the abridged output below, all

    calls originating from a 5-digit extension beginning with 52 (i.e., 52XXX) will be prefixed with

    732852, and a 10 digit calling party number of the form 7328522XXX will be sent, when any ofthe SIP trunk groups (30, 31, 32, 60, 61, 62) in the configuration are chosen for the call.

    Although not shown, similar configuration covered other telephone extension ranges, such as51XXX.

    change public-unknown-numbering 0 Page 1 of 2NUMBERI NG - PUBLI C/ UNKNOWN FORMAT

    TotalExt Ext Tr k CPN CPNLen Code Gr p( s) Pref i x Len

    Total Admi ni st ered: 135 5 5 Maxi mumEnt r i es: 99995 52 30 732852 10

    5 52 31 732852 10

    5 52 32 732852 10

    5 52 60 732852 10

    5 52 61 732852 105 52 62 732852 10

    3.8. Configure ARS Analysis For Outbound Call Routing

    Location-based routing is configured so that users at different locations that dial the same

    telephone number can have calls choose different route-patterns and trunks. In the sampleconfiguration, users at the primary site that dial PSTN telephone numbers will preferentially use

    trunks at the primary site. Similarly, users at the secondary site that dial PSTN telephone

    numbers will preferentially use trunks at the secondary site. Upon congestion or failure, calls

    can use the alternate sites trunks.

    The following screen shows a sample ARS configuration for location 1. If a user at location 1,

    such as extension 51003, dials the ARS access code followed by 1-732-852-XXXX, the call willselect route pattern 60.

    change ars analysis 1732 location 1 Page 1 of 2ARS DI GI T ANALYSI S TABLE

    Locat i on: 1 Per cent Ful l : 1

    Di al ed Tot al Rout e Cal l Node ANISt r i ng Mi n Max Pat t ern Type Num Reqd

    1732852 11 11 60 natl n

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    The following screen shows a sample ARS configuration for location 3. If a user at location 3,

    such as extension 52020, dials the ARS access code followed by 1-732-852-XXXX, the call willselect route pattern 30.

    change ars analysis 1732 location 3 Page 1 of 2ARS DI GI T ANALYSI S TABLE

    Locat i on: 1 Per cent Ful l : 1

    Di al ed Tot al Rout e Cal l Node ANISt r i ng Mi n Max Pat t ern Type Num Reqd

    1732852 11 11 30 natl n

    3.9. Configure Incoming Call Handl ing Treatment For Inbound DigitManipulation

    The incoming call handling treatment for a trunk group can be used to manipulate the digits

    received for an incoming call. In the sample configuration, the number sent from the (simulated)SIP Service Provider has no direct relationship to the corresponding extension in Communication

    Manager. Therefore, all digits are deleted, and the desired Communication Manager

    extension is inserted. In the sample configuration, if a PSTN user dials 732-852-1816, thenumber 21816 arrives via one of the SIP Trunks (as shown in Figure 2). The incoming callhandling table maps 21816 to 52020, the local extension corresponding to the external PSTN

    number. During testing, the number to insert was varied, so that different types of telephones

    (e.g., IP, digital) and vector directory numbers (VDN) could be tested.

    In reference [JRR], the incoming call handling treatment for trunks 30, 31, 60, and 61 arepresented. The two new SIP trunk groups associated with the Processor Ethernet SIP signaling

    groups are similarly configured, as shown below.

    change inc-call-handling-trmt trunk-group 32 Page 1 of 30

    I NCOMI NG CALL HANDLI NG TREATMENTSer vi ce/ Number Number Del I nser tFeature Len Di gi t s

    public-ntwrk 5 21816 all 52020

    public-ntwrk 5 22940 all 51003

    The corresponding configuration for trunk group 62 is shown below.

    change inc-call-handling-trmt trunk-group 62 Page 1 of 30I NCOMI NG CALL HANDLI NG TREATMENT

    Ser vi ce/ Number Number Del I nser tFeature Len Di gi t s

    public-ntwrk 5 21816 all 52020

    public-ntwrk 5 22940 all 51003

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    3.10. Announcement-Related Configuration

    Section 3.12 of reference [JRR] describes the announcement-related configuration retained in the

    verification of these Application Notes. The verifications in Section 5 include calls requesting

    these announcements.

    3.11. Summarizing VDN and Vector-Related Configuration

    Section 3.13 of reference [JRR] describes the VDN and vector-related configuration retained inthe verification of these Application Notes. The verifications in Section 5 include calls to VDNs.

    3.12. Summarizing ESS-Related Configuration

    Section 3.14 of reference [JRR] describes the ESS-related configuration retained in the

    verification of these Application Notes. Product documentation [ESS] should be consulted for

    more information on configuring ESS.

    3.13. Saving Conf iguration Changes

    The command save translation all can be used to save the configuration. In the sample

    configuration, translations were automatically saved and synchronized with each survivable

    processor, such as the ESS, on a nightly basis, as a result of the bold parameters in the screenshown below.

    change system-parameters maintenance Page 1 of 3MAI NTENANCE- RELATED SYSTEM PARAMETERS

    OPERATI ONS SUPPORT PARAMETERSCPE Al arm Act i vat i on Level : mi nor

    SCHEDULED MAI NTENANCEStart Time: 22 : 00

    Stop Ti me: 06 : 00Save Translation: daily

    Update LSP and ESS Servers When Saving Translations: y

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    4. Configure Acme Packet Net-Net Session DirectorsThis section describes only the changes to the configuration of the Session Directors, compared

    to the configuration fully documented in reference [JRR]. Appendix A of reference [JRR]

    contains the output of the show running-configuration command from the primary site Session

    Director. For additional details on the configuration of the Session Director, refer to [AP1].

    The Session Director can be configured via the Acme Packet Command Line Interface (ACLI).This section assumes the reader is familiar with accessing and configuring the Session Director.

    Figure 4 on the following pageillustratesa pictorial view of the sample configuration for the

    Acme Packet Net-Net 4500 at the primary site. Although specifics such as IP addresses will

    vary, the configurations for the Acme Packet Net-Net 4500 at the primary and secondary sitesare conceptually identical, unless noted otherwise. Figure 4is identical to Figure 3 from

    reference [JRR] except that the shaded session-agent with the IP Address of the active S8730

    Server has been added to the session-group ENTERPRISE. The configuration of the Session

    Director at the secondary site is similar, but of course the appropriate IP Addresses shown inFigure 1must be substituted, including the new session-agent with the IP Address of the

    Processor Ethernet of the ESS.

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    Figure 4: Pictorial View of the Primary Site Session Director Configuration

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    4.1. Acme Packet Command Line Interface Summary

    The Session Director is configured using the Acme Packet Command Line Interface (ACLI).

    The following are the generic ACLI steps for configuring various elements.

    1.

    Access the console port of the Session Director using a PC and a terminal emulationprogram such as HyperTerminal. Use the following settings for the serial port on the PC.

    Bits per second: 115200

    Data bits: 8

    Parity : None

    Stop bits: 1

    Flow control: None

    2. Log in to the Session Director with the user password.

    3. Enable the Superuser mode by entering the enable command and then the superuserpassword. The command prompt will change to include a # instead of a > while in

    Superuser mode. This level of system access (i.e., at the acmesystem# prompt) will be

    referred to as themainlevel of the ACLI. Specific sub-levels of the ACLI will then beaccessed to configure specific elements and specificparametersof those elements.

    4. In Superuser mode, enter the configure terminal command. The configure terminal

    command is used to access the system level where all operating and system elements maybe configured. This level of system access will be referred to as theconfigurationlevel.

    5. Enter the name of an element to be configured (e.g., system).

    6. Enter the name of a sub-element, if any (e.g., phy-interface).

    7. Enter the name of an element parameter followed by its value (e.g., name s0p0).8. Enter doneto save changes to the element. Use of the donecommand causes the system

    to save and display the settings for the current element.

    9. Enter exit as many times as is necessary to return to the configuration level.

    10.

    Repeat Steps 4 - 8to configure all the elements.11. Enter exitto return to the main level.

    12. Type save-config to save the entire configuration.13. Type activate-config to activate the entire configuration.

    After accessing different levels of the ACLI to configure elements and parameters, it is necessaryto return to the main level to run certain tasks such as saving the configuration, activating the

    configuration, or rebooting the system.

    4.2. System Conf iguration

    The system configuration defines system-wide parameters for the Session Director. The system

    configuration documented in Section 4.2 of reference [JRR] applies; no changes are necessarydue to the use of Processor Ethernet SIP trunks in Communication Manager.

    4.3. Physical and Network Interfaces

    The physical configuration documented in Section 4.3 of reference [JRR] applies. In the sample

    configuration, for each Session Director, the Ethernet interface slot 0 / port 0 was connected to

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    the external untrusted network, and Ethernet slot 1 / port 0 was connected to the internal

    corporate LAN. A network interface was defined for each physical interface to assign it aroutable IP address.

    Section 6 documents an issue where the Avaya ESS will expect a response to ICMP ping from

    the far-end of SIP trunks, upon activation and control by an ESS running version 5.2 plus SP1.Although this is a problem to be resolved in a subsequent software load, the commands below

    allow a work-around to be implemented on the Acme Packet Session Director at the secondary

    site. In the output below, the only changes to the configuration documented in reference [JRR]are shown in bold. In short, this allows signaling group 62 to remain in-service after an ESS

    takes over, if the ESS is running version 5.2 SP1. See Section 5.7.1.2 for verification of this

    work-around for signaling group 62.

    **sbcsecsi t e- pr i # conf i g t**sbcsecsi t e- pri ( conf i gure)# systemnet work- i nt er f ace**sbcsecsi t e- pri ( net work- i nt er f ace) # sel ect: :1: wancom1: 0 i p= gw=

    2: wancom2: 0 i p= gw=3: s0p0: 0 i p=10. 3. 3. 145 gw=10. 3. 3. 14: s1p0: 0 i p=2. 2. 185. 145 gw=2. 2. 185. 1

    sel ect i on: 4**sbcsecsi t e- pri ( net work- i nt er f ace) # add-hip-ip 2.2.185.145**sbcsecsi t e- pri ( net work- i nt er f ace) # add-icmp-ip 2.2.185.145**sbcsecsi t e- pri ( net work- i nt er f ace) # donenet work- i nt er f ace

    name s1p0sub- port - i d 0descri pt i onhost namei p- address 2. 2. 185. 145pri - ut i l i t y-addr 2. 2. 185. 146sec-uti l i t y-addr 2. 2. 185. 147net mask 255. 255. 255. 0

    gat eway 2. 2. 185. 1

    hip-ip-list 2.2.185.145f t p- addr essicmp-address 2.2.185.145

    **sbcsecsi t e- pri ( net work- i nt er f ace) # exi t**sbcsecsi t e- pri ( system) # exi t**sbcsecsi t e- pr i (conf i gure)# exi t**sbcsecsi t e- pri # savechecki ng conf i gur ati onSave- Conf i g r ecei ved, pr ocessi ng.wai t i ng f or request t o f i ni shRequest t o ' SAVE- CONFI G' has Fi ni shed,Save compl et eCur r ent l y act i ve and saved conf i gur ati ons do not match!

    To sync & act i vat e, r un ' act i vat e- conf i g' or ' r eboot act i vat e' .*sbcsecsi te-pr i # act i vat e- conf i gActi vat e-Conf i g recei ved, pr ocessi ng.wai t i ng f or request t o f i ni shRequest t o ' ACTI VATE- CONFI G' has Fi ni shed,Act i vate Compl et e

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    4.4. Realm

    A realm represents a group of related Session Director components. Defining realms allowsflows to pass through a connection point between two networks. Two realms were defined for

    the interoperability test. The OUTSIDErealm was defined for the external network and theINSIDErealm was defined for the internal network. The realm configuration documented in

    Section 4.4 of reference [JRR] applies; no changes are necessary due to the use of ProcessorEthernet SIP trunks in Communication Manager.

    4.5. SIP Configuration

    The SIP configuration (sip-config) defines global system-wide SIP parameters. The SIP

    configuration documented in Section 4.5 of reference [JRR] applies; no changes are necessarydue to the use of Processor Ethernet SIP trunks in Communication Manager.

    4.6. SIP Interface

    The SIP interface (sip-interface) defines the receiving characteristics of the SIP interfaces on the

    Session Director. Two SIP interfaces were defined, one for each realm. The SIP interface

    configuration documented in Section 4.6 of reference [JRR] applies; no changes are necessary

    due to the use of Processor Ethernet SIP trunks in Communication Manager.

    4.7. Session Agent

    A session agent defines the characteristics of a signaling peer to the Session Director. All the

    session-agents previously configured in reference [JRR] continue to apply. In these Application

    Notes, a session agent for the Processor Ethernet of the active S8730 Server is added to theprimary site Session Director. Similarly, a session agent for the Processor Ethernet of the S8500

    ESS is added to the secondary site Session Director. Key session agent (session-agent)

    parameters include:

    hostname: Fully qualified domain name or IP address of this SIP peer. ip-address: The IP address of this SIP peer.

    port: The port used by the peer for SIP traffic.

    app-protocol: SIP

    transport-method: StaticTCP. With static TCP, a TCP connection can be re-used for

    multiple sessions. With the alternative DynamicTCP, a new connection must beestablished for each session. DynamicTCP also works with Communication Manager,

    but since DynamicTCP had already been documented in the compliance testingperformed in reference [AC-HA], static TCP was used in this sample configuration to

    show that it is a viable option for interoperability with Communication Manager.

    realm-id: The realm id where this peer resides.

    description: A descriptive name for the peer. max-sessions: Although not used in the sample configuration, this parameter can allow

    call admission control to be applied for the session agent. For example, in reference

    [JSR], the max-sessions parameter is configured to match the number of members in thecorresponding Avaya SIP trunk group. If this is not configured, and the Session Director

    sends an INVITE to an Avaya signaling group whose corresponding trunk group has no

    available members, Communication Manager will respond with a SIP 503. The Session

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    Director will redirect the call to another session agent in the SAG. In the context of a

    session agent group whose members are comprised of session agents with differentcapabilities, the max-sessions parameter could be used to effect changes to the

    distribution of sessions across the various agents.

    ping-method: OPTIONS;hops=0 The SIP OPTIONS message will be sent to the peer to

    verify that the SIP connection is functional. In addition, this parameter causes the SessionDirector to set the Max-Forward field to 0 in outbound OPTIONS pings generated by the

    Session Director to this session-agent.

    ping-interval: Specifies the interval between SIP OPTIONS pings in seconds. Since

    the intent is to monitor the health of the connection, pings may be suppressed if there is

    traffic to/from the session-agent that shows the connection is up.

    ping-in-service-response-codes: Although not configured as part of the base

    verification testing, this parameter can be used to specify the list of response codes that

    keep a session agent in-service. By default, any response from the session agent is

    enough to keep the session agent in service. If it is desired that only a 200 OK response

    is a valid response to OPTIONS to mark the session agent in-service, then 200 can be

    entered. Note that Communication Manager will respond to OPTIONS with a 503 invarious conditions where the SIP trunk group corresponding to the SIP signaling group

    would not be able to process a call. Examples include when the SIP trunk group isadministratively busied out, or when the trunk group is in-service, but there are no

    available resources to handle a call (e.g. all trunk members in use for calls). Other cases

    where the Processor Ethernet SIP trunks will respond with a 503 are documented in theverifications in Section 5.

    out-service-response-codes Although not defined in the sample configuration, this

    parameter can be used to specify the list of OPTIONS ping response codes that take a

    session agent out-of-service.

    options trans-timeouts=1 This parameter defines the number of consecutive non-ping

    transaction timeouts that will cause the session agent to be marked out-of-service. Forexample, with this option set to 1, if an INVITE is sent to an Avaya session agent that is

    currently marked in-service, but no response is received resulting in a transactiontimeout, the session agent will be immediately marked out-of-service. In the sample

    configuration, where session agent groups are used, this allows future calls to flow to in-

    service session agents in the group without experiencing a delay due to a transactiontimeout. Note that an explicit error response, such as a 503, is not considered a

    transaction timeout.

    reuse-connections TCP Enables TCP connection re-use.

    tcp-keepalive enabled Enables standard TCP Keep-Alives

    tcp-reconn-interval 10 Specifies the idle time, in seconds, before TCP keep-alive

    messages are sent.

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    The key settings for the session agent to the active S8730 Server are shown below. This

    configuration is added on the Acme Packet Session Director at the primary site only.

    The key settings for the session agent to the ESS are shown below. This configuration is

    added on the Acme Packet Session Director at the secondary site only.

    sessi on- agenthostname 2.2.185.88ip-address 2.2.185.88

    port 5060st ate enabl edapp-protocol SIP

    app- t ypetransport-method StaticTCPrealm-id INSIDE

    egr ess- r eal m- i ddescription ESS-PE

    carr i ersal l ow- next - hop- l p enabl edconst r ai nt s di sabl edmax- sessi ons 0< t ext r emoved f or br evi t y >

    ping-method OPTIONS;hops=0ping-interval 16

    pi ng-send-mode keep- al i veping-in-service-response-codesout-service-response-codesoptions trans-timeouts=1

    reuse-connections TCPtcp-keepalive enabledtcp-reconn-interval 10< t ext r emoved f or br evi t y >

    sessi on- agenthostname 2.2.87.13ip-address 2.2.87.13

    port 5060st ate enabl edapp-protocol SIP

    app- t ypetransport-method StaticTCPrealm-id INSIDE

    egr ess- r eal m- i ddescription S8730-PE-Active-IP

    carr i ersal l ow- next - hop- l p enabl edconst r ai nt s di sabl edmax- sessi ons 0< t ext r emoved f or br evi t y >

    ping-method OPTIONS;hops=0ping-interval 16

    pi ng-send-mode keep- al i veping-in-service-response-codes

    out-service-response-codesoptions trans-timeouts=1

    reuse-connections TCPtcp-keepalive enabledtcp-reconn-interval 10< t ext r emoved f or br evi t y >

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    4.8. Session Agent Groups (SAG)

    A session agent group is a logical collection of two or more session agents that behave as asingle aggregate entity. In the sample configuration documented in reference [JRR], the two C-

    LANs at each site are configured in a session agent group within the Acme Packet Net-Net 4500

    at that site. Although not required, a session agent group is also created for connectivity to the

    (simulated) SIP PSTN, for easy adaptation to outside networks with multiple options for nexthops.

    For the sample configuration associated with these Application Notes, the session agentsassociated with the Avaya Processor Ethernet (defined in Section 4.7) are added. In the

    primary site Session Director, the session agent for the active Avaya S8730 Server is added to

    the SAG ENTERPRISE. In the secondary site Session Director, the session agent for theS8500 ESS is added to the SAG ENTERPRISE.

    Key session group (session-group) parameters include:

    group-name: a unique name for the session agent group.

    app-protocol: SIP strategy: selects the algorithm to use for distribution of traffic among the session agents

    in the group. In the sample configuration, a simple roundrobin distribution is selectedfor alternating traffic among the session agents. More sophisticated alternatives are

    available including leastbusy and propdist (Proportional Distribution) based on

    configurable session constraints, and are described in reference [AP1].

    dest: The IP Addresses of the session agents that are members of the session agent group.

    sag-recursion If enabled, allows re-trying another session agent in the session agent

    group after a failure for the previously selected session agent. For example, if anINVITE message is sent to a session agent, and the session agent does not respond, SAG

    recursion allows an INVITE to be automatically directed to another session agent in the

    SAG. Those familiar with Communication Manager terminology may benefit from thefollowing parallel. Conceptually, SAG recursion is similar to Avaya Look-Ahead

    Routing, where the SAG is the route-pattern, the session agents are the trunk groups in

    the route-pattern and SAG recursion allows LAR to the next trunk in the pattern upon afailure. Note that this analogy is imperfect in that the Acme Packet Net-Net Session

    Director can make decisions about which session-agent in the SAG to choose based onalgorithms that would check usage and load before selecting the next session agent. In

    the sample configuration, SAG recursion is enabled for the session agent group

    containing the interfaces to Communication Manager.

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    Since the session agent group ENTERPRISE already exists, and the only desired change is

    for the dest parameter to include the Processor Ethernet entry, the following approach canbe used to change only the dest. The commands shown below were issued on the Session

    Director at the primary site. The IP Addresses of the destination session agents can be

    included inside parenthesis, separated by spaces, as shown in bold below.

    acmesbc- pr i # conf i g tacmesbc- pr i ( conf i gur e) # sessi on- r out eracmesbc- pr i ( sessi on- r out er) # sessi on- gr oupacmesbc- pr i ( sessi on- agent - gr oup) # sel ect:1: SERV_PROVI DER2: ENTERPRI SE

    sel ecti on: 2

    acmesbc- pr i ( sessi on- agent - gr oup) # dest (2.2.85.2 2.2.85.20 2.2.87.13)acmesbc- pri ( sessi on- agent - gr oup)# showsessi on- gr oup

    gr oup- name ENTERPRI SEdescri pt i onst ate enabl edapp- pr otocol SI Pst r at egy RoundRobi ndest

    2. 2. 85. 22. 2. 85. 202. 2. 87. 13

    t r unk- gr oupsag- r ecur si on enabl edst op- sag- r ecur se 401, 407

    acmesbc- pr i ( sessi on- agent - gr oup) #acmesbc- pr i ( sessi on- agent - gr oup) # done

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    For the Acme Packet Net-Net 4500 at the secondary site, the configuration can also be

    updated to add the session agent for the S8500 ESS to the existing ENTERPRISE SAG.To show an alternative approach, a different syntax is shown in bold below. Rather than

    including all the destinations within parenthesis, the dest +2.2.185.88 approach was used to

    add the session agent for the ESS to the SAG ENTERPRISE. The before and after

    configuration can be observed in the show output below.

    4.9. SIP Manipulation

    SIP manipulations are rules used to modify the SIP messages. For example, SIP manipulations

    can be performed to ensure private network topology hiding and confidentiality. The SIPmanipulation configuration documented in Section 4.9 of reference [JRR] applies; no changesare necessary due to the use of Processor Ethernet SIP trunks in Communication Manager.

    4.10. Steering Pools

    Steering pools define sets of ports that are used for steering media flows (e.g., RTP) through the

    Acme Packet Net-Net 4500. The selected ports are used to modify the SDP to cause receiving

    sbcsecsi t e- pr i ( sessi on- rout er ) # sessi on- groupsbcsecsi t e- pr i ( sessi on- agent - group) # sel ect:1: SERV_PROVI DER2: ENTERPRI SE

    sel ecti on: 2sbcsecsi t e-pri ( sessi on- agent - gr oup) # showsessi on- gr oup

    gr oup- name ENTERPRI SEdescri pt i onst ate enabl edapp- pr otocol SI Pst r at egy RoundRobi n

    dest2. 2. 185. 22. 2. 185. 20

    t r unk- groupsag- r ecur si on enabl edst op- sag- r ecur se 401, 407l ast - modi f i ed- by admi n@2. 2. 4. 150l ast - modi f i ed- dat e 2009-04- 14 11: 47: 50

    sbcsecsi t e-pri ( sessi on- agent - gr oup) # dest +2.2.185.88sbcsecsi t e-pri ( sessi on- agent - gr oup) # showsessi on- gr oup

    gr oup- name ENTERPRI SEdescri pt i onst ate enabl edapp- pr otocol SI Pst r at egy RoundRobi ndest

    2. 2. 185. 22. 2. 185. 202. 2. 185. 88

    t r unk- groupsag- r ecur si on enabl edst op- sag- r ecur se 401, 407l ast - modi f i ed- by admi n@2. 2. 4. 150l ast - modi f i ed- dat e 2009-04- 14 11: 47: 50

    sbcsecsi t e-pri ( sessi on- agent - gr oup) #sbcsecsi t e-pri ( sessi on- agent - gr oup) # done

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    session agents to direct media to the Acme Packet Net-Net 4500. Two steering pools were

    defined, one for each realm. The steering pools documented in Section 4.10 of reference [JRR]apply; no changes are necessary due to the use of Processor Ethernet SIP trunks in

    Communication Manager. Consult reference [AP1] for more information, including a means to

    use steering pool configuration for call admission control.

    4.11. Local Policy

    Local policy controls the routing of SIP calls from one realm to another. The local policiesdocumented in Section 4.11 of reference [JRR] apply; no changes are necessary due to the use of

    Processor Ethernet SIP trunks in Communication Manager.

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    5. VerificationsThis section illustrates expected behaviors and sample results, focusing on calls using the

    Processor Ethernet SIP trunks. Reference [JRR] can be consulted for a robust set of related call

    verifications using the C-LAN SIP trunks, including failure scenarios. Section 6 documents test

    observations that resulted in product modification requests.

    5.1. Normal Operation

    When all components are functioning normally, calls are processed by the active S8730 Server at

    the primary site. Inbound and outbound calls can use SIP Trunks at either site, subject to routing

    rules and efficient allocation of resources. In normal operation, the Processor Ethernet of theAvaya S8730 Servers can process calls, but the Processor Ethernet of the ESS will not participate

    in SIP call traffic.

    The following screen, taken from the active S8730 Server during normal operation, shows that

    cluster 1, the S8730 Server pair, controls both the primary and secondary site. The Connected

    Clus(ter) IDs shows that cluster 2, the S8500 ESS, can be reached by the IPSIs.

    status ess port-networks

    Cl ust er I D 1 ESS PORT NETWORK I NFORMATI ON

    Por t I PSI Pri / Pr i / Cnt l ConnectedCom I nt f I nt f Nt wk Gt way Sec Sec Cl us Cl us( t er )

    PN Num Loc Type Ste Loc Loc St ate I D I Ds1 1 1A01 IPSI up 1A01 1A01 actv-aa 1 1 2

    2 2 2A01 IPSI up 2A01 2A01 actv-aa 1 1 2

    The following screen, taken from the active S8730 Server during normal operation, shows that

    the primary and secondary site IPSIs are controlled by the S8730 Server and in-service".

    list ipserver-interface

    I P SERVER I NTERFACE I NFORMATI ON

    Por t Pr i / Pr i mar y/ Pr i mar y/ Pr i mar y/ St at e OfNt wk Sec Secondary Secondary Secondary Serv Cont r ol Heal t hNum Bd Loc I P Address Host Name DHCP I D State St at e C P E G- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -

    1 1A01 2. 2. 185. 9 2. 2. 185. 9 i psi - A01a IN act v- aa 0. 0. 0. 0

    2 2A01 2. 2. 85. 9 2. 2. 85. 9 i psi - A02a IN actv- aa 0. 0. 0. 0

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    The following screen, taken during normal operation, shows that the ESS with IP Address

    2.2.185.88 is registered with up-to-date translations.

    list survivable-processor

    SURVIVABLE PROCESSORS

    Name Type IP Address Reg Act Translations Net

    Updated RgnESSCid002Sid003 ESS S 2.2.185.88 y n 22:00 6/18/2009 1

    S83LSP- i n- G250 LSP 2. 2. 25. 88 y n 22: 00 6/ 18/ 2009 2S83LSP- i n- G700 LSP 2. 2. 1. 88 y n 22: 00 6/ 18/ 2009 4

    In normal operation, when all network elements are functioning, and the active Avaya S8730

    is processing all calls, incoming and outgoing SIP trunk calls to the enterprise can use SIP trunkgroups 30, 31, 32, 60, or 61. Table 2shows the expected states during normal operation. Trunk

    group 62 will not be used during normal operation because Communication Manager trunk

    group 62 will be out-of-service (OOS). Also, the secondary site Acme Packet session agent to

    the ESS will be out-of-service.

    Avaya TrunkGroup (TG) &

    Service State via

    Avaya S8730

    Servers

    Reason forAvaya

    Service State

    Primary SiteNN4500

    Session Agent

    & Service

    State

    SecondarySite NN4500

    Session Agent

    & Service

    State

    Reason forAcme Packet

    Service State

    Where

    Applicable

    30 (C-LAN)in-service

    OK responseto OPTIONS

    SA 2.2.85.2In-service

    N/A Any SIP response toOPTIONS

    31 (C-LAN)

    in-service

    OK response

    to OPTIONS

    SA 2.2.85.20

    In-service

    N/A Any SIP response to

    OPTIONS

    32 (S8730 PE)

    in-service

    OK response

    to OPTIONS

    SA 2.2.87.13

    In-service

    N/A Any SIP response to

    OPTIONS

    60 (C-LAN)in-service

    OK responseto OPTIONS

    N/A SA 2.2.185.2In-service

    Any SIP response toOPTIONS

    61 (C-LAN)in-service

    OK responseto OPTIONS

    N/A SA 2.2.185.20In-service

    Any SIP response toOPTIONS

    62 (S8730 PE)

    Out-of-service /Far-End

    S8730 PE is

    not a sessionagent in the

    secondary site

    NN4500

    N/A SA 2.2.185.88

    Out-of-Service

    No SIP response from

    ESS at secondary site,ESS not active

    Table 2: Normal Operation, All Network Elements Functioning

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    During normal operation, signaling group 62 will be marked for bypass, as shown below. The

    Session Director at the secondary site does not respond at the SIP level to the OPTIONS messagesent by the active S8730 Server procr, the near-end of signaling group 62.

    status signaling-group 62

    STATUS SI GNALI NG GROUP

    Gr oup I D: 62 Act i ve NCA- TSC Count : 0Gr oup Type: si p Act i ve CA- TSC Count : 0

    Si gnal i ng Type: f aci l i t y associ at ed si gnal i ngGroup State: far-end bypass

    During normal operation, the members of trunk group 62 will all be out-of-service/far-end asshown in the abridged output below. Signaling group 62 and trunk group 62 will only be in-

    service when the ESS is active (e.g., controlling a port network or gateway).

    status trunk 62 Page 1

    TRUNK GROUP STATUS

    Member Por t Service State Mt ce Connect ed Port sBusy

    0062/ 001 T00140 OOS/FE-idle no0062/ 002 T00141 OOS/FE-idle no0062/ 003 T00142 OOS/FE-idle no0062/ 004 T00143 OOS/FE-idle no

    5.1.1. Incoming Calls from PSTN Arriving via SIP Trunk to Primary Site

    Using the sample configuration, incoming PSTN calls arriving via the primary site will bedistributed in a round-robin fashion to the three session agents in the SAG ENTERPRISE.

    These include the two C-LAN session agents previously illustrated in reference [JRR] as well as

    the new session agent corresponding to the Processor Ethernet of the active S8730 Server.

    The following show sipd agent command was run on the primary Acme Packet Net-Net 4500

    after three inbound PSTN calls via the primary site Session Director. All session agents are in-

    service. Session agent 10.3.3.40 (the SIP Service Provider) shows 3 Active Inbound sessions.The three session agents that are members of the SAG ENTERPRISE each show one Active

    Outbound session, as a result of the round-robin distribution.

    acmesbc-pri# show sipd agent

    10:27:29-34 (recent)

    ----- Inbound ----- ---- Outbound ----- -- Latency -- Max

    Session Agent Active Rate ConEx Active Rate ConEx Avg Max Burst

    10.3.3.40 I 3 0.0 0 0 0.0 0 0.010 0.012 1

    2.2.85.2 I 0 0.0 0 1 0.0 0 0.118 0.118 1

    2.2.85.20 I 0 0.0 0 1 0.0 0 0.118 0.118 12.2.87.13 I 0 0.0 0 1 0.0 0 0.002 0.003 1

    The following trace output shows a call incoming on signaling group 32 / trunk group 32 fromPSTN telephone 732-852-2550. The incoming call handling table for trunk group 32 mapped the

    received number (21816) to extension 52020. Extension 52020 is an IP Telephone with IP

    Address 2.2.1.109 in Region 3. Initially, the IP Media Processor in region 3 (2.2.26.4) is used,

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    but as can be seen in the final trace output, once the call is answered, the final RTP media path is

    ip-direct from the IP Telephone (2.2.1.109) to the inside of the Acme Packet Net-Net 4500at the primary site (2.2.85.45).

    list trace tac 132 Page 1LI ST TRACE

    t i me data10: 59: 36 Cal l i ng par t y t r unk- group 32 member 1 ci d 0xa9710: 59: 36 Cal l i ng Number & Name 7328522496 NO- CPName10: 59: 36 act i ve t r unk- group 32 member 1 ci d 0xa9710: 59: 36 di al 5202010: 59: 36 r i ng st at i on 52020 ci d 0xa9710: 59: 36 G711MU ss: of f ps: 20

    r gn: 3 [ 2. 2. 1. 109] : 19036r gn: 3 [ 2. 2. 26. 4] : 13828

    10: 59: 36 G711MU ss: of f ps: 20r gn: 3 [ 2. 2. 85. 45] : 49160r gn: 3 [ 2. 2. 26. 4] : 13820

    10: 59: 42 act i ve st at i on 52020 ci d 0xa9710: 59: 42 G711MU ss: of f ps: 20

    r gn: 3 [ 2. 2. 85. 45] : 49160

    r gn: 3 [ 2. 2. 1. 109] : 1903610: 59: 42 G711MU ss: of f ps: 20

    r gn: 3 [ 2. 2. 1. 109] : 19036r gn: 3 [ 2. 2. 85. 45] : 49160

    The following portion of a Wireshark trace shows an incoming call using the S8730 ProcessorEthernet. In frame 49, the Service Provider delivers the inbound INVITE to the Session Director

    at the primary site. In Frame 51, the Session Director sends the INVITE to the S8730 Processor

    Ethernet. The call proceeds as usual using the S8730 Processor Ethernet.

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    In the next example, a call arrives via the Processor Ethernet SIP trunk (32), and the call is

    directed to call vector 1 via a VDN (x51081) that plays an announcement (x22232), and thencollects and verifies password digits from a caller. This type of call verifies proper collection of

    DTMF via RFC 2833, and also illustrates the Communication Manager audio group concept that

    allows announcements to be sourced from the local Avaya gateway. Audio groups enable

    efficient utilization of resources, and also provide redundancy benefits.

    From the bolded rows, note that the tone receiver as well as the announcements are sourced from

    a board in the 2A carrier at the primary site.

    list trace tac 132 Page 1LI ST TRACE

    t i me data11: 05: 34 Cal l i ng par t y t r unk- group 32 member 1 ci d 0xa9f11: 05: 34 Cal l i ng Number & Name 7328522496 NO- CPName11: 05: 34 act i ve t r unk- group 32 member 1 ci d 0xa9f11: 05: 34 di al 5108111: 05: 34 r i ng vect or 1 ci d 0xa9f11: 05: 34 G711MU ss: of f ps: 20

    r gn: 3 [ 2. 2. 85. 45] : 49172r gn: 3 [ 2. 2. 26. 4] : 13956

    11: 05: 36 tone-receiver 02AXX04 ci d 0xa9f11: 05: 36 act i ve announcement 22232 ci d 0xa9f11: 05: 36 hear audio-group 1 board 02A08 ext 22232 ci d 0xa9f11: 05: 50 act i ve announcement 22234 ci d 0xa9f11: 05: 50 hear audio-group 1 board 02A08 ext 22234 ci d 0xa9f

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    In the next example, a call arrives via the Processor Ethernet site SIP trunk (32), and the call is

    directed to Avaya call vector 3 via a VDN (x51082) that plays an announcement (x22555) andcollects digits for call routing. This also shows the locally-sourced announcements, verifies

    DTMF collection using RFC 2833, and illustrates a call that arrives via the primary site

    Processor Ethernet SIP trunk, but connects with a user (x51003) at the secondary site.

    list trace tac 132 Page 1LI ST TRACE

    t i me data11: 09: 13 Cal l i ng par t y t r unk- group 32 member 1 ci d 0xaa311: 09: 13 Cal l i ng Number & Name 7328522496 NO- CPName11: 09: 13 act i ve t r unk- group 32 member 1 ci d 0xaa311: 09: 13 di al 5108211: 09: 13 r i ng vect or 3 ci d 0xaa311: 09: 13 G711MU ss: of f ps: 20

    r gn: 3 [ 2. 2. 85. 45] : 49178r gn: 3 [ 2. 2. 26. 4] : 14040

    11: 09: 15 t one- r ecei ver 02B0108 ci d 0xaa311: 09: 15 act i ve announcement 22555 ci d 0xaa311: 09: 15 hear audi o- group 1 board 02A08 ext 22555 ci d 0xaa3

    11: 09: 26 di al 5100311: 09: 26 r i ng st at i on 51003 ci d 0xaa311: 09: 26 G711MU ss: of f ps: 20

    r gn: 1 [ 2. 2. 185. 200] : 3472r gn: 3 [ 2. 2. 26. 4] : 14048

    11: 09: 31 act i ve st at i on 51003 ci d 0xaa311: 09: 31 G711MU ss: of f ps: 20

    r gn: 3 [ 2. 2. 85. 45] : 49178r gn: 1 [ 2. 2. 185. 200] : 3472

    11: 09: 31 G711MU ss: of f ps: 20r gn: 1 [ 2. 2. 185. 200] : 3472r gn: 3 [ 2. 2. 85. 45] : 49178

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    5.1.2. Outgoing Call to PSTN from Primary Site

    The following trace shows an outbound ARS call from IP Telephone x52020 to the PSTNnumber 17328522550. The call is routed based on the location of the originator to route pattern

    30, which contains trunk group 32. The call initially uses a media processor in region 3

    (2.2.26.4), but after the call is answered, the call is shuffled to become an ip-directconnection between the IP Telephone (2.2.1.109) and the Acme Packet Session Director at the

    primary site (2.2.85.45).

    list trace tac 132 Page 1LI ST TRACE

    t i me data16: 25: 53 di al 9917328522550 r out e: ARS16: 25: 53 r out e- patt ern 30 pr ef erence 1 ci d 0xa0b16: 25: 53 sei ze t r unk- group 32 member 2 ci d 0xa0b16: 25: 53 Set up di gi t s 732852255016: 25: 53 Cal l i ng Number & Name 7328522020 J ohn Publ i c16: 25: 53 Pr oceed t r unk- group 32 member 2 ci d 0xa0b16: 25: 53 G711MU ss: of f ps: 20

    r gn: 3 [ 2. 2. 85. 45] : 49154r gn: 3 [ 2. 2. 26. 4] : 2764

    16: 25: 59 act i ve t r unk- group 32 member 2 ci d 0xa0b16: 25: 59 G711MU ss: of f ps: 20

    r gn: 3 [ 2. 2. 1. 109] : 58096r gn: 3 [ 2. 2. 85. 45] : 49154

    16: 25: 59 G711MU ss: of f ps: 20r gn: 3 [ 2. 2. 85. 45] : 49154r gn: 3 [ 2. 2. 1. 109] : 58096

    The following portion of a Wireshark trace shows an outgoing call using the S8730 Processor

    Ethernet. In frame 1670, Communication Manager uses the S8730 Processor Ethernet to send anINVITE to the primary site Session Director. In frame 1673, the Session Director sends the

    INVITE on to the Service Provider. The call proceeds as usual using the S8730 ProcessorEthernet.

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    Outbound calls were also made to PSTN destinations requiring a log-in with password, such as a

    messaging system, to verify that DTMF was working properly in the outbound direction. Thefollowing is an example Communication Manager trace using the Processor Ethernet SIP trunk

    62. The call is placed from an H.323 IP Telephone (x52020). When the call is answered, the

    call is shuffled to ip-direct, from the IP Telephone (2.2.1.109) to the primary site Session

    Director (2.2.85.45). After the user hears the prompt for password input, and begins to enterdigits at 10:07:18, the media shuffles back to the Media Processor serving region 3 at 2.2.26.4.

    Once user digit entry ends, and the user in this case is listening to a voice message, the media

    shuffles back to ip-direct, as can be seen at 10:07:56. This behavior is not unique to calls usingProcessor Ethernet SIP Trunks.

    list trace tac 132 Page 1LI ST TRACE

    t i me data10: 07: 13 di al 9917328523500 r out e: ARS10: 07: 13 r out e- patt ern 30 pr ef erence 1 ci d 0xa8110: 07: 13 sei ze t r unk- group 32 member 13 ci d 0xa8110: 07: 13 Set up di gi t s 732852350010: 07: 13 Cal l i ng Number & Name 7328522020 J ohn Publ i c10: 07: 13 Pr oceed t r unk- group 32 member 13 ci d 0xa8110: 07: 13 act i ve t r unk- group 32 member 13 ci d 0xa8110: 07: 13 G711MU ss: of f ps: 20

    r gn: 3 [ 2. 2. 85. 45] : 49154r gn: 3 [ 2. 2. 26. 4] : 13388

    10: 07: 13 G711MU ss: of f ps: 20r gn: 3 [ 2. 2. 1. 109] : 19036r gn: 3 [ 2. 2. 85. 45] : 49154

    10: 07: 13 G711MU ss: of f ps: 20r gn: 3 [ 2. 2. 85. 45] : 49154r gn: 3 [ 2. 2. 1. 109] : 19036

    10: 07: 18 G711MU ss: of f ps: 20r gn: 3 [ 2. 2. 1. 109] : 19036r gn: 3 [ 2. 2. 26. 4] : 13416

    10: 07: 18 G711MU ss: of f ps: 20r gn: 3 [ 2. 2. 85. 45] : 49154r gn: 3 [ 2. 2. 26. 4] : 13424

    10: 07: 56 G711MU ss: of f ps: 20r gn: 3 [ 2. 2. 1. 109] : 19036r gn: 3 [ 2. 2. 85. 45] : 49154

    10: 07: 56 G711MU ss: of f ps: 20r gn: 3 [ 2. 2. 85. 45] : 49154r gn: 3 [ 2. 2. 1. 109] : 19036

    5.1.3. Incoming Calls from PSTN Arriving Via PSTN to Secondary Site

    Using the sample configuration, incoming PSTN calls arriving via the secondary site will bedistributed in a round-robin fashion to the session agents in the SAG ENTERPRISE defined in

    the secondary site Session Director. These include the two C-LAN session agents previouslyverified in reference [JRR] as well as the new session agent corresponding to the Processor

    Ethernet of the ESS. Under normal operation, as noted previously, the session agentcorresponding to the Processor Ethernet of the ESS will be out-of-service from the point of view

    of the secondary site Session Director. Therefore, incoming calls from the PSTN arriving via the

    secondary site are identical to those shown in Section 5.1.3 of reference [JRR].

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    41 of 70

    CM-PE-NN4500

    To illustrate the behavior, the following show sipd agent command was run on the secondary

    Acme Packet Net-Net 4500 after three inbound PSTN calls via the secondary site. Observe thatsession agent 2.2.185.88 is out of service. Session agent 10.3.3.1 (the SIP Service Provider)

    shows 3 Active Inbound sessions. Session agent 2.2.185.2 show two Active Outbound sessions,

    and session agent 2.2.185.20 shows one Active Outbound session, as a result of the round-robin

    distribution, and the bypass of the out-of-service session agent in the SAG ENTERPRISE.

    sbcsecsite-pri# show sipd agent

    11:33:28-56 (recent)

    ----- Inbound ----- ---- Outbound ----- -- Latency -- Max

    Session Agent Active Rate ConEx Active Rate ConEx Avg Max Burst

    10.3.3.1 I 3 0.0 0 0 0.0 0 0.004 0.005 1

    2.2.185.2 I 0 0.0 0 2 0.0 0 0.098 0.118 1

    2.2.185.20 I 0 0.0 0 1 0.0 0 0.118 0.118 1

    2.2.185.88 O 0 0.0 0 0 0.0 0 0.000 0.000 0

    5.1.4. Outgoing Call to PSTN from Secondary Site

    During normal operation, outgoing calls to the PSTN via the secondary site are identical to those

    shown in Section 5.1.4 of reference [JRR]. Since the Processor Ethernet of the S8500 ESS is notchosen for outbound SIP trunk calls during normal operation, there is no change to the results

    compared to reference [JRR].

    5.2. Enterprise IP WAN Failure Isolating Secondary Site (ESS ControlsSecondary Site Only)

    This section illustrates behavior when the enterprise IP network that allows the secondary site tocommunicate with other sites is down. Refer to Induced Failure Reference Number 1 in

    Figure 1. Since the secondary site is isolated from the primary site, the ESS in the secondary

    site controls the Avaya G650 Media Gateway in the secondary site. The primary site has not

    experienced a failure, and therefore the primary site active S8730 Server still controls the

    primary site Avaya G650 Media Gateway.

    The following screen, taken from the ESS, shows that the ESS is controlling port network 1 only.

    list ipserver-interface

    I P SERVER I NTERFACE I NFORMATI ON

    Por t Pr i / Pr i mar y/ Pr i mar y/ Pr i mar y/ St at e OfNt wk Sec Secondary Secondary Secondary Ser v Cont r ol Heal t hNum Bd Loc I P Addr ess Host Name DHCP I D Stat e Stat e C P E G- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -

    1 1A01 2.2.185.9 2.2.185.9 ipsi-A01a IN actv-aa 0.0.0.0

    2 2A01 2.2.85.9 2.2.85.9 ipsi-A02a OUT active 0.1.1.0

  • 8/10/2019 Acme Sbc Config

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    Table 3shows the expected states from the point of view of the active Avaya S8730 Server,

    when the WAN is down, and the primary site is controlled by the active Avaya S8730 Server,and the secondary site is controlled by the Avaya ESS. Incoming and outgoing SIP trunk calls at

    the primary site can use SIP trunk groups 30, 31, and 32.

    Avaya TrunkGroup (TG) &

    Service State

    via Avaya

    S8730 Server

    Reason forAvaya

    Service State

    via S8730

    Server

    Primary SiteNN4500

    Session

    Agent &

    Service State

    Secondary SiteNN4500

    Session Agent

    & Service

    State

    Reason forAcme Packet

    Service State

    Where

    Applicable

    30 (C-LAN)

    in-service

    OK response

    to OPTIONS

    SA 2.2.85.2

    In-service

    N/A Any SIP response to

    OPTIONS

    31 (C-LAN)

    in-service

    OK response

    to OPTIONS

    SA 2.2.85.20

    In-service

    N/A Any SIP response to

    OPTIONS

    32 (S8730 PE)

    in-service

    OK response

    to OPTIONS

    SA 2.2.87.13

    In-service

    N/A Any SIP response to

    OPTIONS

    60 (C-LAN)Out-of-service /

    Near-End

    S8730 notcontrolling

    C-LANscabinet

    N/A SA 2.2.185.2In-service

    Any SIP response toOPTIONS

    61 (C-LAN)Out-of-service /

    Near-End

    S8730 notcontrolling

    C-LANs

    cabinet

    N/A SA 2.2.185.20In-service

    Any SIP response toOPTIONS

    62 (S8730 PE)

    Out-of-service /Far-End

    No respon


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