Abstract. Testing was conducted in the Avaya Solution and Interoperability Test Lab, utilizing a test account on Gamma Telecom s production service.

Transcription

1 Avaya Solution & Interoperability Test Lab Application Notes for Configuring SIP Trunk Connectivity between Avaya Meeting Exchange Enterprise Edition 5.2, Avaya Aura Session Border Controller 6.0 and the Gamma Telecom IP Direct Connect Service Issue 1.0 Abstract These Application Notes describe the steps to configure Avaya Meeting Exchange Enterprise Edition 5.2 and Avaya Aura Session Border Controller 6.0 with the Gamma Telecom IP Direct Connect Service. Gamma Telecom IP Direct Connect Service allows PSTN callers to participate in conference calls hosted on a SIP enabled Avaya Meeting Exchange Enterprise Edition 5.2 server without the need for TDM media gateways and the associated maintenance costs. Testing was conducted in the Avaya Solution and Interoperability Test Lab, utilizing a test account on Gamma Telecom s production service. 1 of 36

2 1. Introduction These Application Notes describe the steps to configure Avaya Meeting Exchange Enterprise Edition 5.2 and Avaya Aura Session Border Controller 6.0 with the Gamma Telecom IP Direct Connect Service. The Gamma Telecom IP Direct Connect Service allows PSTN callers to participate in conference calls hosted on a SIP enabled Avaya Meeting Exchange Enterprise Edition 5.2 server without the need for TDM media gateways and the associated maintenance costs. Avaya Meeting Exchange Enterprise 5.2 runs on a single Avaya S8800 server and is connected to Avaya Aura Session Border Controller 6.0 over a TCP-based SIP trunk while DTMF is transmitted within the RTP stream using RFC2833 compliant messages. Avaya Bridge Talk was used to schedule conferences on the bridge, monitor the status of the conference participants and use operator features such as transfer, dial out and hang up. Both Avaya Meeting Exchange Enterprise 5.2 and the Gamma Telecom IP Direct Connect Service support G711A-law and G729 codecs. The hosted Avaya Conferencing test site consists of Avaya Aura Session Border Controller 6.0 and Avaya Meeting Exchange Enterprise 5.2. Avaya Aura Session Border Controller provides topology hiding without the need for Network Address Translation (NAT), SIP header manipulation and SIP signaling and media channel conversion services. The service offer described in these Application Notes is designed for business customers using Avaya Meeting Exchange Enterprise and Avaya Aura Session Border Controller on a private protected IP network who wish to provide their customers and partners with publicly routable E.164 conference access numbers, that could be used to access the conference bridge from the Gamma Telecom IP Direct Connect Service through the Internet. It is assumed that the Avaya Aura Session Border Controller (AASBC) acts as a peering host between the public Internet and the private IP network and provides Denial-of-Service (DoS), packet filtering and topology hiding without the need for an additional firewall or intrusion prevention system (IPS) on either the public or private side of the Avaya Aura Session Border Controller. Hardware, software resilience and failover between the various Avaya components is not covered in these Application Notes. 2 of 36

3 1.1. Reference Configuration Figure 1 illustrates how the Avaya Aura Session Border Controller 6.0 and Avaya Meeting Exchange Enterprise 5.2 were used for the Interoperability testing. The reference configuration is comprised of a sample hosting site connected via a leased line to the Internet. At the edge of the hosting site an Avaya Aura Session Border Controller acts as a B2BUA for SIP calls. The Avaya Aura Session Border Controller terminates and re-originates calls using its own IP addresses thereby hiding the IP address range (topology) of the private network. Network security is provided by the DoS and packet filtering module of the Avaya Aura Session Border Controller. The Avaya Aura Session Border Controller converts the SIP signaling channel from UDP to TCP for inbound and vice-versa for outbound calls. Figure 1: Avaya Interoperability Test Lab Reference Configuration 3 of 36

4 2. Equipment and Software Validated The following equipment and software were used in the reference configuration. Equipment Software Version Avaya S8800 Server Avaya Aura Session Border Controller R (GA) Avaya S8800 Server Avaya Meeting Exchange Enterprise Edition R5.2 Build (GA) GENBAND S3 Session Firmware 4.3 Border Controller Avaya S8800 Server Avaya Aura Session Border Controller R (GA) Test PC1 Microsoft Windows Vista with Avaya BridgeTalk (GA) Table 1: Equipment and Software Used in the Reference Configuration 4 of 36

5 3. Configure Avaya Aura Session Border Controller This section provides the procedures for configuring Session Border Controller and includes the following items: Log in to Avaya Aura Session Border Controller using the GUI Verify Outside Interface Settings Verify Inside Interface Settings Configure SIP Header Manipulation Rules Configure SIP Gateways Configure Dial Plan Save and Activate the Configuration These Application Notes assume that the Session Border Controller was installed as per [1] Log in to Avaya Aura Session Border Controller using the GUI Configuration is accomplished by accessing the browser-based GUI of Session Border Controller, using the URL where <ip-address> is the IP address of either the inside or outside interface of the Session Border Controller. Log in with the appropriate credentials. 5 of 36

6 3.2. Verify Outside Interface Settings The Session Border Controller uses the outside interface to communicate with the Gamma Telecom SIP peer and needs to have a publicly routable IP address assigned to one of its virtual interfaces in order to be able to reach the Gamma Telecom SIP peer ( ). Note that the physical outside interface of the Session Border Controller (eth2) may have many virtual outside interfaces (eth2:x) assigned to it. The following steps verify the IP addressing, routing and SIP configuration of the outside interface. The Home page is displayed. Select the Configuration tab on the toolbar. 6 of 36

7 The Configuration Loaded page is displayed. Expand box -> interface eth2 and click on ip togamma. The Configure cluster\box:aasbc.avaya.com\interface eth2\ip togamma page is displayed. Verify that the interface has a publicly routable IP address with the correct subnet mask. 7 of 36

8 Scroll down to the sip section and verify that a udp-port is enabled. This port is used for sending and receiving SIP messages to/from the Gamma SIP peer. Scroll down to the routing section and verify that an IP route exists to the Gamma SIP peer. 8 of 36

9 3.3. Verify Inside Interface Settings The Session Border Controller uses the inside interface to communicate with Meeting Exchange and has a private RFC 1918 IP address assigned to it. The following steps verify the IP addressing, routing and SIP configuration of the inside interface. From the left pane expand interface eth0, click ip inside. The Configure cluster\box:aasbc.avaya.com\interface eth0\ip inside page is displayed. Verify that the interface has a private IP address with the correct subnet mask. Scroll down to the sip section and verify that a tcp-port is enabled. This port is used for sending and receiving SIP messages to/from the Meeting Exchange. 9 of 36

10 In the sample configuration Meeting Exchange was installed on the same private IP subnet as the inside interface of the Session Border Controller therefore an IP route is not required Configure SIP Header Manipulation Rules SIP header manipulation rules are required to convert the transport type from TCP to UDP for outbound calls, as the Gamma SIP peer supports only SIP over UDP requests. Also, the host part of the Request-URI, From and To headers would have to be modified before the call leaves the outside interface of the Session Border Controller as the Gamma SIP peer expects its own IP address in the host part of these headers. Expand vsp session-config-pool. The Configure vsp\session-config-pool page is displayed. Click Add entry. 10 of 36

11 The Create vsp\session-config-pool\entry - Step 1 of 1: Edit entry page is displayed. In the name field type a descriptive name then click Create. The Configure vsp\session-config-pool\entry togamma page is displayed (not shown). Scroll down to the uri section. Click Configure next to the to-uri-specification element. 11 of 36

12 The Configure vsp\session-config-pool\entry togamma\to-uri-specification page is displayed. In the host field enter the IP address of the Gamma SIP peer. Select UDP from the transport drop-down list box. Click Set to commit the changes. The Configure vsp\session-config-pool\entry togamma page is displayed. Repeat the previous step for the from-uri-specification and request-uri-specification elements. Click Configure next to the from-uri-specification element. 12 of 36

13 The Configure vsp\session-config-pool\entry togamma\from-uri-specification page is displayed. In the host field enter the IP address of the Gamma SIP peer. Select UDP from the transport drop-down list box. Click Set to commit the changes. The Configure vsp\session-config-pool\entry togamma page is displayed. Click Configure next to the request-uri-specification element. 13 of 36

14 The Configure vsp\session-config-pool\entry togamma\request-uri-specification page is displayed. In the host field enter the IP address of the Gamma SIP peer. Select UDP from the transport drop-down list box. Click Set to commit the changes Configure SIP Gateways The steps of adding two SIP Gateways to the Session Border Controllers configuration are included in this section. One SIP Gateway object is required for Meeting Exchange and another for the Gamma SIP peer. From the left pane, expand vsp then click on enterprise. 14 of 36

15 The Configure vsp\enterprise page is displayed. Scroll down then click Add sip-gateway. The Create vsp\enterprise\servers\sip-gateway - Step 1 of 1: Edit sip-gateway page is displayed. In the name field type a descriptive name for the Gamma SIP peer then click Create. 15 of 36

16 The Configure vsp\enterprise\servers\sip-gateway Gamma page is displayed. Under servers: server-pool click Configure. The Configure vsp\enterprise\servers\sip-gateway Gamma.\server-pool page is displayed. Click Add server. 16 of 36

17 The Create vsp\enterprise\servers\sip-gateway Gamma.\server-pool\server - Step 1 of 1: Edit server page is displayed. In the server-name text field type a descriptive name for the Gamma SIP peer. In the host text field type the IP address of the Gamma SIP peer. Click Create. The Configure vsp\enterprise\servers\sip-gateway Gamma.\server-pool\server Gamma page is displayed. Under transport select UDP from the drop-down list box. Verify that under port 5060 is set (default). Click Set to commit the changes. 17 of 36

18 From the left pane expand vsp enterprise. Click servers. The Configure vsp\enterprise\servers page is displayed. 18 of 36

19 Scroll down and at the bottom of the page click Add sip-gateway. The Create vsp\enterprise\servers\sip-gateway - Step 1 of 1: Edit sip-gateway page is displayed. In the name field type a descriptive name for Meeting Exchange. Click Create. 19 of 36

20 The Configure vsp\enterprise\servers\sip-gateway MeetingExchange page is displayed. Under servers: server-pool select Configure. The Configure vsp\enterprise\servers\sip-gateway MeetingExchange\server-pool page is displayed.click Add server then Set to commit the changes. 20 of 36

21 The Create vsp\enterprise\servers\sip-gateway MeetingExchange\server-pool\server - Step 1 of 1: Edit server page is displayed. Under server-name type a descriptive name. Under host type the IP address of Meeting Exchange. Click Create. 21 of 36

22 3.6. Configure Dial Plan In the sample configuration two dial plan entries were administered. An outbound entry for routing calls from Meeting Exchange to a PSTN Operator and an inbound entry for routing calls from PSTN users to the conference bridge. From the left pane expand vsp and click dial-plan. The Configure vsp\dial-plan page is displayed. In the route section click Add route. The Create vsp\dial-plan\route - Step 1 of 1: Edit route page is displayed. In the name field type a descriptive name then click Create. 22 of 36

23 The Configure vsp\dial-plan\route togamma page is displayed. Under request-uri-match -> type select phone-prefix from the drop-down list box. In the phone-prefix field type the PSTN phone number of the Operator. This is the number that Meeting Exchange will dial when Audio Path to Operator is activated in BridgeTalk. Meeting Exchange will send a SIP INVITE request with the PSTN phone number of the Operator in the Request-URI and To headers. Under minimal-digits enter the number of digits that the Session Border Controller should try to match from the user part of the incoming Request-URI or select as-is if you d like to match on the full called number. Scroll down till the peer section. Under type select server from the drop-down list box. Select Configure vsp\enterprise\servers\sip-gateway Gamma from the server drop-down list box. 23 of 36

24 Scroll down to the bottom of the page. From the session-config drop-down list box select vsp\session-config-pool\entry togamma we created is Section 3.4. Click Set to commit the changes. In the sample configuration the conference access number was The following dial plan rule was created to compare the received user part of the incoming Request-URI header to a phone-prefix entry. If a match occurs the called number gets translated to 4444 which was the DNIS on Meeting Exchange. This step is optional. DNIS value can be up to 16 digits long, however for ease of administration, some deployments tend to use shorted (4/5 digit) conference access numbers (DNIS). Expand vsp then click dial-plan. The Configure vsp\dial-plan page is displayed. Scroll down and click Add route. 24 of 36

25 The Create vsp\dial-plan\route - Step 1 of 1: Edit route page is displayed. In the name text field type a descriptive name for the dial plan rule that will translate the long incoming number to Click Create. The Configure vsp\dial-plan\route GammatoMX page is displayed. Under request-uri-match, select phone-prefix from the type drop-down list box. In the phone-prefix text field type the DNIS that PSTN participants would dial to access the conference. Under minimal-digits select asis from the drop-down list box. Scroll down. In the peer section select server from the type drop-down list box. Select vsp\enterprise\servers\sip-gateway MX from the server drop-down list box. Click Set to commit the changes. 25 of 36

26 Scroll down. Under request-user select replace-prefix from the drop-down list box. In the newphone-prefix text field type the DNIS administered on Meeting Exchange. In the sample configuration Repeat these steps for the to-user section. Click Set to commit the changes Save and Activate the Configuration From the left pane select Configuration Update and save configuration. 26 of 36

27 The following popup screen appears. Click OK. Click OK at the next popup window. The Configuration Updated and Saved page is displayed. 27 of 36

28 4. Configure Avaya Meeting Exchange Enterprise This section provides the procedures for configuring Meeting Exchange and includes the following items: Configure SIP Listener Configure Dialout Configure DNIS Size (optional) Configure DNIS Mappings Configure Codecs Restart the Conference Bridge It is assumed, that Meeting Exchange is installed, configured and licensed as per [2]. The following instructions also assume the user is logged in to the Meeting Exchange Application Server Linux console using SSH Configure SIP Listener Configure the following settings to enable SIP connectivity on the Meeting Exchange server: Edit /usr/ipcb/config/system.cfg using the Linux vi tool or download the file to your local machine using a Secure Copy Protocol (SCP) client (i.e.: WinSCP) for editing. o Add the IP address of the Meeting Exchange server: IPAddress= as shown below. o Add a line to populate the From Header Field in SIP INVITE messages. The following SIP URI will be displayed when the conference operator calls a participant: MyListener=<sip:6000@ :5060;transport=tcp> o Add a line to provide a SIP Device Contact address to use for acknowledging SIP messages: respcontact=<sip:6000@ :5060;transport=tcp> # ip address of the server IPAddress= # request we will be listening to MyListener=<sip:6000@ :5060;transport=tcp> # if this setting is populated will Overwrite the contact field in responses respcontact=<sip:6000@ :5060;transport=tcp> 28 of 36

29 4.2. Configure Dialout Edit /usr/ipcb/config/telnumtouri.tab file with a text editor on Meeting Exchange Enterprise Application server. Add the following line to the file to route outbound calls from the Avaya Meeting Exchange Enterprise server to external PSTN numbers. Note that the host part of the Request-URI contains the IP address of the inside interface of the Session Border Controller as it is the next-hop for all SIP messages sent from Meeting Exchange. * sip:$0@ :5060;transport=tcp 4.3. Configure DNIS Size (optional) The DNIS is the number that the users dial to access a conference. In the sample configuration PSTN users dial an 11 digit DNIS ( ) to access the conference bridge. An inbound dial plan rule from Section 3.6 translates into To increase the DNIS size to 11 digits, run the cbutil utility on Meeting Exchange as follows: [mx52-a ~]# cbutil dnissize 11 cbutil Copyright 2004 Avaya, Inc. All rights reserved. 29 of 36

30 4.4. Configure DNIS Mappings To map DNIS entries to a conference, run the cbutil utility on Meeting Exchange as follows: [mx52-a ~]# cbutil add N SCAN [mx52-a ~]# cbutil add N SCAN At the command prompt, enter cbutil list to verify the DNIS entries provisioned. [mx52-a ~]# cbutil list cbutil Copyright 2004 Avaya, Inc. All rights reserved. DNIS Grp Msg PS CP Function On Failure Line Name Company Name Room Start Room End N SCAN ENTER N SCAN ENTER Configure Codecs By default only G.711A-law and G.711Mu-law is enabled on Meeting Exchange while G.729 is disabled. Edit /usr/ipcb/config/audiopreferences.cfg file with a text editor on Meeting Exchange Enterprise Application server to enable G.729 and disable G711Mu-law as it is not supported by the Gamma SIP peer, by deleting/adding the # sign at the beginning of the line. # audiopreferences.cfg # This table is an ordered list of MIME subtypes specifying the codecs supported # by this media server. The list is specified in the order in which an SDP offer # will list the various MIME subtypes on the m=audio line. # For static payload type numbers (i.e. numbers between 0-96) please use the # iana registered numbering scheme. # See: mimesubtype payloadtype Performance #PCMU 0 10 PCMA 8 10 #G G of 36

31 4.6. Restart the Conference Bridge After the configuration changes are made, restart the Meeting Exchange Enterprise Application Server: Log in to the MX Application Server using the dcbmaint account. Issue the dcbmaint command. The System Maintenance Main Menu screen is displayed. Navigate to Re-Initialization. 31 of 36

32 Press Enter and at the prompt type yes. The dcbmaint utility terminates and the following message is displayed: 32 of 36

33 5. Verification Steps The Session Border Controller stores the SIP signaling traces of each test call in the Call Log database. Log in to the Session Border Controller through the GUI and click on Call Logs. 33 of 36

34 Calls can be filtered by Call ID and called/calling number. Click on Detail once a particular call is selected to display the SIP message trace (not shown) Troubleshooting Tools A SIP protocol analyzer such as Wireshark can be used to capture SIP traces at the various interfaces. SIP traces can be instrumental in understanding SIP protocol issues resulting from configuration problems. 6. Conclusion As illustrated in these Application Notes Avaya Meeting Exchange Enterprise 5.2 and Avaya Aura Session Border Controller R6 can be configured to interoperate successfully with the Gamma Telecom IP Direct Connect Service. This solution provides PSTN users the ability to participate in conference calls hosted in a secure data centre over a public SIP trunk using the Gamma Telecom IP Direct Connect Service. The reference configuration shown in these Application Notes is representative of a basic enterprise customer configuration and is intended to provide configuration guidance to supplement other Avaya product documentation. 34 of 36

35 7. References The Avaya product documentation is available at unless otherwise noted. [1] Application Notes for Configuring Avaya AuraTM Communication Manager with Avaya AuraTM Session Border Controller 6.0 for Gamma Telecom IP Direct Connect SIP Trunks - Issue 1.0 [2] Administering Meeting Exchange Servers Release 5.2.1, December 11, 2009, of 36

36 Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by and are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes. Please any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya Solution & Interoperability Test Lab at 36 of 36