Mitel SIP CoE Technical Configuration

Transcription

1 Mitel SIP CoE Technical Configuration G12 Communications 1 (877) sales@g12com.com 150 Lake Street South, Kirkland, WA 98033

2 NOTICE The information contained in this document is believed to be accurate in all respects but is not warranted by Mitel Networks Corporation (MITEL ). The information is subject to change without notice and should not be construed in any way as a commitment by Mitel or any of its affiliates or subsidiaries. Mitel and its affiliates and subsidiaries assume no responsibility for any errors or omissions in this document. Revisions of this document or new editions of it may be issued to incorporate such changes. No part of this document can be reproduced or transmitted in any form or by any means - electronic or mechanical - for any purpose without written permission from Mitel Networks Corporation. TRADEMARKS Mitel is a trademark of Mitel Networks Corporation. Windows and Microsoft are trademarks of Microsoft Corporation. Other product names mentioned in this document may be trademarks of their respective companies and are hereby acknowledged. Mitel Technical Configuration Notes: Configure MiVoice Business 7.2 to use with G12 Communications SIP trunks November , Trademark of Mitel Networks Corporation Copyright 2016, Mitel Networks Corporation All rights reserved ii

3 Table of Contents OVERVIEW... 1 Interop History... 1 Interop Status... 1 Software & Hardware Setup... 1 Tested Features... 2 Device Limitations and Known Issues... 3 Network Topology... 4 CONFIGURATION NOTES... 5 MiVoice Business Configuration Notes... 5 Network Requirements... 5 Assumptions for MiVoice Business Programming... 5 Licensing and Option Selection SIP Licensing... 6 Class of Service Options... 7 Network Elements Network Element Assignment (Outbound Proxy) Trunk Attributes SIP Peer Profile SIP Peer Profile Assignment by Incoming DID ARS Digital Modification Plans ARS Routes ARS Digits Dialed Fax Configuration Network Zones MiVoice Border Gateway Configuration Notes iii

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5 Overview This document provides a reference to Mitel Authorized Solutions providers for configuring the MiVoice Business (MiVB) 7.2 to connect to the G12 Communications Sip trunk via MBG. Different components can be configured in various configurations depending on your VoIP solution. This document covers a basic setup with basic SIP trunk routing rules. Interop History Version Date Reason 1 September, 2016 Initial Interop with Mitel MiVoice Business release 7.2 Load and G12 Communications SIP trunk. Interop Status This Interop of G12 Communications SIP trunk with MiVoice Business 7.2 has been given a Compatible Certification status. This SIP trunk will be included in the SIP CoE Reference Guide. The most common certification which means MiVoice Business has been tested and/or validated by the Mitel SIP CoE team. Product support will provide all necessary support related to the interop, but issues unique or specific to the 3rd party will be referred to the 3rd party as appropriate. Software & Hardware Setup The table below provides the hardware and software specifications used to generate SIP Audio calls, both point to point and conference calls, using G12 Communications SIP trunk connected to MiVoice Business 7.2. Manufacturer Variant Software Version Mitel MiVoice Business 7.2 Mxe-III Platform Release 7.2, Mitel MiVoice Business Gateway (MBG) MBG: Mitel MBG-Teleworker MSL: MBG: Mitel MiCollab NuPoint (UM and speech AA) MiCollab: Mitel 68xx series SIP sets Mitel 53xx series IP sets G12 Communications G12 Communications NetSapiens SAS Media Server 12.25x NetSapiens SAS Session Border Controller 12.25x 1

6 Tested Features Below table provides an overview of the features tested during the Interoperability test cycle and not a detailed view of the test cases. Please see the SIP Trunk Side Interoperability Test Plan for detailed test cases and results (APTest session 700) Feature Feature Description Issues Basic Call Authentication Packetization rate Session Timers Automatic Call Distribution Making and receiving a call through SIP service provider and their PSTN gateway, call holding, transferring, conferencing, busy calls, long calls durations, variable codec Making and receiving authenticated calls. Making and receiving basic calls with different packetization rates. Long outgoing and incoming basic calls with and without setting session timer. Making calls to an ACD environment with RAD treatments, Interflow and Overflow call scenarios and DTMF detection. NuPoint Voic AWC Personal Ring Groups (PRG) Teleworker Codec Video Fax Fax Terminating calls to a NuPoint voic boxes and DTMF detection. Making AWC conference with outbound and inbound callers. Receiving calls through MiVoice Business and their PSTN gateway to a personal ring group. Also moving calls to/from the prime member and group members. Making and receiving a call through MiVoice Business and their PSTN gateway to and from Teleworker extensions. All testcases were performed using G.711 codec. Basic G.729 incoming/outgoing call, local/remote hold were performed. Making and receiving a call through MiVoice Business with video capable devices. Fax calls using G.711 codec. T.38 Fax calls. - No issues found - Issues found, cannot recommend to use - Issues found Not tested/applicable 2

7 Device Limitations and Known Issues This is a list of problems or not supported features when G12 Communications SIP trunk is connected to the MiVoice Business 7.2 Feature Supervised Transfer Authentication Problem Description During testing it was discovered that an inbound call which is then supervised transferred back to the network will have no audio. Unsupervised transfers work correctly. Recommendation: Please contact G12 Communications for more information on a resolution for this issue. G12 Communications does not support authentication. Recommendation: Please contact G12 Communications for more information. T.38 FAX G12 Communications uses a custom configuration for T.38 FAX calls. Recommendation: Contact G12 Communications for details. 3

8 Network Topology This diagram shows how the testing network is configured for reference. Figure 1 Network Topology 4

9 Configuration Notes This section is a description of how the SIP Interop network was configured. These notes provide a guideline as how a device can be configured in a customer environment and how the G12 Communications and MiVB were configured in our test environment. Disclaimer: Although Mitel has attempted to setup the interop testing facility as closely as possible to a customer premise environment, implementation setup could be different onsite. YOU MUST EXERCISE YOUR OWN DUE DILIGENCE IN REVIEWING, planning, implementing, and testing a customer configuration. MiVoice Business Configuration Notes The following information shows how to configure a MiVoice Business 7.2 to interconnect with G12 Communications SIP trunk. Network Requirements There must be adequate bandwidth to support the VoIP network. As a guide, the Ethernet bandwidth is approx 85 Kb/s per G.711 voice session and 29 Kb/s per G.729 voice session (assumes 20ms packetization). As an example, for 20 simultaneous SIP sessions, the Ethernet bandwidth consumption will be approx 1.7 Mb/s for G.711 and 0.6Mb/s. Almost all Enterprise LAN networks can support this level of traffic without any special engineering. Please refer to the MiVoice Business Engineering guidelines on the Mitle edocs Website ( for further information. For high quality voice, the network connectivity must support a voice-quality grade of service (packet loss <1%, jitter < 30ms, one-way delay < 80ms). Assumptions for MiVoice Business Programming The SIP signaling connection uses UDP on Port

10 Licensing and Option Selection SIP Licensing Ensure that MiVoice Business is equipped with enough SIP trunking licenses for the connection to G12 Communications. This can be verified within the License and Option Selection form. Enter the total number of licenses in the SIP Trunk Licences field. This is the maximum number of SIP trunk sessions that can be configured in the MiVB to be used with all service providers, applications and SIP trunking devices. Figure 2 License and Option Selection 6

11 Class of Service Options The Class of Service Options form is used to create or edit a Class of Service and specify the associated options. Classes of Service, identified by Class of Service numbers, are referenced in the Trunk Attributes form for SIP trunks. Navigation: System Properties-> System Feature Settings-> Class of Service Options The Class of Service Options Assignment form is used to create or edit a Class of Service and specify its options. Classes of Service, identified by Class of Service numbers, are referenced in the Trunk Service Assignment form for SIP trunks. Figure 3 Class of Service 7

12 Class of Service for Trunk General Figure 4 Class of Service (Basic) for SIP Trunk 8

13 Figure 5 Class of Service (Basic) for SIP Trunk cont. 9

14 Figure 6 Class of Service (Basic) for SIP Trunk cont. 10

15 Figure 7 Class of Service (Basic) for SIP Trunk cont. Figure 8 - Class of Service (Basic) for SIP Trunk cont. 11

16 Advanced Figure 9 Class of Service (Advanced) for SIP Trunk 12

17 Figure 10 Class of Service (Advanced) for SIP Trunk cont. 13

18 Network Elements Create a network element for a SIP Peer, G12 Communications. In this example, the gateway is reachable by an IP Address and is defined as G12 Communications in the Network Elements form. The FQDN or IP addresses of the SIP Peer (Network Element), the External SIP Proxy, and Registrar are provided by your service provider. If your service provider trusts your network connection by asking for your gateway external IP address, then programming the IP address for the SIP Peer, Outbound Proxy and Registrar is not required for SIP trunk integration. This will need to be verified with your service provider. Set the transport to UDP and port to Figure 11 Network Elements 14

19 Network Element Assignment (Outbound Proxy) In addition, depending on your configuration, an Outbound Proxy may need to be configured to route SIP data to the service provider. If you have a Proxy server installed in your network, MiVoice Business will require knowledge of this by programming the Proxy as a network element then referencing this proxy in the SIP Peer profile assignment (later in this document). Figure 12 Network Element Assignment (Proxy) 15

20 Trunk Attributes Use Trunk Attributes form to configure Trunk Service Number. In this example, the Trunk Service Number 17 will be used to direct incoming calls to an answer point in MiVoice Business. Program the Non-dial In or Dial In Trunks (DID) according to the site requirements and what type of service was ordered from your service provider. Figure 13 below shows an example of a configuration for incoming DID calls. The MiVoice Business will absorb 3 digits from the incoming digit string leaving the remaining digits for routing purposes. The remaining digits must be a dialable number within the MiVb. Please refer to MiVoice Business 7.2 System Administration documentation for further programming information. Figure 13 Trunk Service Assignment 16

21 SIP Peer Profile The recommended connectivity via SIP Trunking does not require additional physical interfaces. IP/Ethernet connectivity is part of the base MiVoice Business Platform. The SIP Peer Profile should be configured as shown in Figures Under Basic tabs: Network Element: The selected SIP Peer Profile needs to be associated with previously created G12 Communications Network Element. Registration User Name: The MiVoice Business does not support Bulk registration. As such trunks have to be registered individually. As G12 Communications did not require registration this field was left blank. Address Type: Select IP address of the MiVoice Business as shown. Outbound Proxy Server: Select Network Element previously configured for the outbound proxy server. Trunk Service: Enter the trunk service number previously configured, 9 for this configuration. SMDR: If Call Detail Records (CDR) are required for SIP Trunking, the SMDR Tag should be configured (by default there is no SMDR and this field is left blank). Maximum Simultaneous Calls: Configure this entry to be the maximum number of SIP trunks provided by G12 Communications. Authentication: Enter the username and password provided by the provider. Please note that G12 Communications did not require Authentication thus did not provide any username and password. NOTE: Ensure the remaining SIP Peer profile policy options are similar to the screen capture below. 17

22 Figure 14 SIP Peer Profile - Basic Figure 15 SIP Peer Profile Call Routing 18

23 Under Calling Line ID tab, enter the value for the Default CPN as provided by G12 Communications and leave the rest as default settings. Figure 16 SIP Peer Profile Calling Line ID 19

24 Under SDP Options tab: Figure 17 SIP Peer Profile SDP Options 20

25 Under Signaling and Header Manipulation tab: G12 Communications SIP does not support PRACK. As such, we recommend that option Disable Reliable Provisional Responses is set to Yes, and Require Reliable Provisional Responses on Outgoing Calls is set to No. Figure 18 SIP Peer Profile Signaling and Header Manipulation 21

26 For Timers, the Session timer was increased to Figure 19 SIP Peer Profile Timers For Key Press Event, Outgoing DID Ranges, and Profile Information tabs, leave the default settings intact. 22

27 SIP Peer Profile Assignment by Incoming DID This form is used to associate DID range of numbers from G12 Communications SIP Trunk to a particular SIP Peer profile. Enter one or more telephone numbers. The maximum number of digits per telephone number is 26. You can enter a mix of ranges and single numbers. The entire field width is limited to 60 characters. In this particular example (Figure 20 SIP Peer Profile Assignment by Incoming DID) we configured 4 DID numbers of 10 digits assigned to this trunk, they represent the agents desk phone extensions. G12 Communications will be using these numbers to reach the MCD. Use a comma to separate telephone numbers and ranges. Use a dash (-) to indicate a range of telephone numbers. The first and last characters cannot be a comma or a dash. If the numbers do not fit within the 60 characters maximum, you can create a new entry for the same profile. Use a '*' to reduce the number of entries that need to be programmed. This is a type of "prefix identifier", and cannot be used as a range with '-'. For example, the string "11*" would be used to associate a peer with any number in the range from 110 up to the maximum digits per telephone number (In this case, ) Note that the string "11" by itself would not count as a match, as the '*' represents 1 or more digits. Figure 20 SIP Peer Profile Assignment by Incoming DID 23

28 ARS Digital Modification Plans Ensure that ARS Digit Modification for outgoing calls on the SIP trunk to G12 Communications absorbs or inject additional digits according to your dialing plan. In this example, we will be absorbing 3 digits, for example 971 prefix to dial out. Figure 21 ARS Digit Modification Plans 24

29 ARS Routes Create a route for SIP Trunks connecting a trunk to G12 Communications. In this example, the SIP trunk is assigned to Route Number 18. Choose SIP Trunk as a routing medium and choose the SIP Peer Profile and Digit Modification entry created earlier. Figure 22 ARS Routes 25

30 ARS Digits Dialed ARS initiates the routing of trunk calls when certain digits are dialed from a station. In this example, when a user dials 950 followed by 11 digits, the call will be routed out of the MiVB via Route 18 which was defined in the previous step. Figure 23 ARS Digit Dialed 26

31 Fax Configuration G12 Communications uses the inter-zone FAX profile communication (Profile 2). The Fax Service Profiles form allows user to define the settings for FAX communication over the IP network. Below default settings can be modified: Inter-zone FAX profile: defines the FAX settings between different zones in the network. There is only one Inter-zone FAX profile; it applies to all inter-zone FAX communication. It defaults to image data transmission protocol V.17 at bps speed. It defines the settings for FAX Relay (T.38) FAX communication. Intra-zone FAX profile: defines the FAX settings within each zone in the network. Profile 1 defines the settings for G.711 pass through communication. Profile 2 to 64 define the settings for FAX Relay (T.38) FAX communication. All zones default to G.711 pass through communication (Profile 1). Figure 24 Fax Service Profile 27

32 Figure 25 Fax Advanced Settings 28

33 Network Zones By default, all zones are set to Intra-zone FAX Profile 1, which defaults to G.711 passthrough communication. Based on your network diagram, assign the Intra-zone FAX Profiles to the Zone IDs of the zones. If audio compression is required within the same zone, set Intra-Zone Compression to Yes. In our example, Zone ID 2 is setup with Intra-zone fax Profile 2 which supports T.38 communication. Figure 26 Network Zones 29

34 MiVoice Border Gateway Configuration Notes When configuring MiVoice Border Gateway (MBG), you need to identify the working MiVB where to forward SIP messages to and then to configure the SIP trunk. To do this: Login to MBG and click MiVoice Border Gateway In right pane, click Service configuration tab and then ICPs On ICPs page, ensure that the working MiVB is configured. If needed, click Plus button link and add a new Mitel switch. Figure 27 MiVB Configuration page Click Save button To add a new SIP trunk: Click Service configuration tab and then click SIP trunking Click Add a SIP trunk link 30

35 Enter the SIP trunk s details as shown in Figure 28: Name is the name of the trunk Remote trunk endpoint address the FQDN of the provider s switch or gateway (this address should be given to you by the provider, e.g. G12 Communications). Local/Remote RTP framesize (ms) is the packetization rate you want to set on this trunk Local Streaming - Should be enabled. RTP address override MBG WAN interface. PRACK Support - disabled Routing rule one it allows routing of any digits to the selected Mitel MiVB The rest of the settings are optional and could be configured if required. Click Save button Figure 28 SIP Trunk configuration settings 31

36 To modify a SIP options: Click System configuration tab and then click Settings Figure 29 SIP Options page Check status: click System status tab and then click SIP Trunks 32

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