Overview
This Application Note is a configuration guide for the Ribbon SBC (Session Border Controller) 1000/2000 when connecting to Cisco Unified Communication Manager (CUCM) and PlusNet SIP Trunk.
The configuration guide supports features outlined in the Microsoft Technet web page:
Introduction
Interoperability compliance testing focuses on verifying inbound and outbound call flows between Ribbon SBC 1000/2000 and Cisco CUCM.
Audience
This technical document provides telecommunications engineers with information for configuring both the Ribbon SBC and the third-party product. Procedures in this document require navigating third-party equipment as well as applying Ribbon SBC Command Line Interface (CLI) commands. To complete the configuration and perform any troubleshooting, the engineer performing the procedures must understand the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP.
Requirements
The following table lists the hardware and software used in the reference configuration.
Test Equipment and Software
Vendor | Equipment | Software Version |
---|
Ribbon Networks | SBC 2000 | V8.0.2 |
Third-party Vendor |
Cisco | Cisco Unified CM Administration | 12.0.1.21900 |
Cisco | Cisco SIP Phone 7841 | sip78xx.11-7-1-17 |
VentaFax | Fax Machine VentaFax | 7.6.243.616 |
Reference Configuration
The following figure serves as a topology for the reference configuration. The figure shows the connectivity between third-party equipment and the Ribbon SBC 1000/2000.
Reference Configuration Topology
Support
For questions about information in this document, contact Ribbon Support in either of the following ways:
Verify License
The interoperability test described in this document requires no special licensing.
1. Security Profile
Select System > Security > SIP Trunk Security Profile
SIP Trunk Security Profile
2. SIP Profile
Select Device > Device Settings > SIP Profile
3. SIP Trunk
Select Device > Trunk > Add New
4. Route Group
Select Call Routing > Route/Hunt > Route Group > Add New
5. Route List
Select Call Routing > Route/Hunt > Route List > Add New
6. Route Pattern
Select Call Routing > Route/Hunt > Route Pattern > Add New
Ribbon SBC 1000/2000 Configuration
The following configuration steps provide an example of how to configure the Ribbon SBC 1000/2000 to interoperate with Skype 2015 and Virgin Media SIP Trunk:
- SIP Profile
- SIP Server
- Media System
- Media Profiles
- Media List
- Remote Authorization Tables
- Signaling Groups
- Transformation
- Call Routing Table
1. SIP Profile
SIP Profiles control how the Ribbon SBC 1000/2000 communicates with SIP devices. The profiles control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags.
Select Settings > SIP > SIP Profiles to access the SIP Profile screen.
The following figures show the default SIP profile used for the Ribbon 1000/2000 used for this configuration effort.
2. SIP Server
SIP Server Tables contain information about the SIP devices connected to the Ribbon SBC 1000/2000.
Select Settings > SIP > SIP Server Tables to access the SIP Server Tables screen.
The entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with each SIP server. The entries also contain links to counters that are useful for troubleshooting, as shown in the following figures.
The Media System Configuration contains system-wide settings for the Media System. Configuring the media system means setting the number of RTP/RTCP port pairs and the starting port.
Select Settings > Media > Media System Configuration to access the Media System configuration screen.
Media Profiles specify the individual voice and fax compression codecs and their associated settings for inclusion into a Media List. Different codecs provide varying levels of compression, allowing the reduction of bandwidth requirements at the expense of voice quality.
Select Settings > Media > Media Profiles.
The following figures illustrate possible media profiles of the voice codecs used for the SBC 1000/2000. The examples are for reference only.
The Media List shows the selected voice and fax compression codecs and their associated settings.
Select Settings > Media > Media List to access the Media List configuration screen.
6. Remote Authorization Tables
Remote Authorization Tables and their entries contain information used to respond to request message challenges by an upstream server. The Remote Authorization Tables on this page appear as options in Creating and Modifying Entries in the SIP Servers (For additional information about Remote Authorization Tables, see the Ribbon online SBC 1000/2000 documentation).
Select Settings > SIP > Remote Authorization Tables to access the Remote Authorization Tables configuration screen.
Remote Authorization Table
7. Signaling Groups
Signaling Groups allow telephony channels to be grouped for routing and shared configuration. These groups are the entity to which calls are routed, and the location from which Call Routes are selected. These are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, Signaling Groups will specify protocol settings and links to server, media, and mapping tables.
Select Settings > Signaling Groups to access the Signaling Groups configuration screens.
Transformation Tables facilitate the conversion of names, numbers, and other fields when routing a call. For example, transformations can convert a public PSTN number into a private extension number, or a SIP address (URI). Every entry in a Call Routing Table requires a Transformation Table.
Select Settings > Transformation to access the Transformation configuration screen.
9. Call Routing Table
Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports, and between protocols (like ISDN to SIP). Call Routing Tables define routes. The use of Call Routing Tables allows for flexible configuration of which calls will be carried, and also how the calls are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message translations, Cause Code Reroutes, Media Lists, and the three types of Signaling Groups (ISDN, SIP and, CAS).
Select Settings > Call Routing Table to access the Call Routing Table configuration screen.
Interoperability Test Results
The following table provides test results for interoperability compliance testing between Ribbon SBC 1000/2000 and CUCM
Interoperability Compliance Test Results
Test Number | Test Scenario | Setup / Test Results | Status | Comment |
---|
4.1 | Registration and authentication (registration mode) | The PBX is able to execute the correct resolution oft he DNS SRV record | Pass |
|
4.3 | Basic call | With the basic call tests, the standard call scenarios and the CLIP/CLIR features are tested. - Only en-bloc dialing is supported, overlap sending is not possible.
| Pass |
|
4.3.1 | Normal call | Outgoing call from PBX to PSTN - En-bloc dialling
- Local area call (without area code); area code must be set by the PBX
- Setting of the correct calling number with all available telephone number blocks
- If two or more locations with different area codes are assigned to one SIP trunk, the number in the PAI header has to be the geographic number which belongs to the users location. This is very important in case of emergency calls, because Plusnet uses the number of the PAI header to route the emergency call to the proper emergency call center.
| Pass |
|
4.3.1.1 | Normal call | The geographic number in the PAI header corresponds to the location 11 of the user | Pass | Is it possible to configure several PAI per SIP trunk? Yes |
4.3.1.2 | Normal call | Display of A-number in B-party CLIP (national PSTN) | Pass |
|
4.3.1.3 | Normal call | Display of A-number in B-party CLIP (international PSTN) | Pass |
|
4.3.1.4 | Normal call | Display of A-number in B-party CLIP (mobile) | Pass |
|
4.3.1.5 | Normal call | Call to mobile Outgoing call to mobile => mobile phone turned off | Pass |
|
4.3.1.6 | Normal call | Suppression of A-number => CUR | Pass |
|
4.3.1.7 | Normal call | Outgoing call from analog extension | Pass |
|
4.3.1.8 | Normal call | Outgoing call (> 5 min.) => PSTN | Pass | Hold connection for 5 minutes => RTP still correct? Yes |
4.3.2 | Normal call | Incoming call from PSTN (national) => PBX - Test all available telephone number blocks
| Pass |
|
4.3.2.1 | Normal call | Display of A-number => CLIP | Pass |
|
4.3.2.2 | Normal call | Incoming call from mobile => PBX Display of A-number => CLIP | Pass |
|
4.3.2.3 | Normal call | Suppression of A-number => CLIR | Pass |
|
4.3.3 | Normal call | Two simultaneous outgoing/incoming calls | Pass |
|
4.3.4 | Normal call | Enabled feature DND (do not disturb) | Pass |
|
4.3.5 | Normal call | Test call with codec G.711 | Pass |
|
4.3.6 | Normal call | Test call with codec G.722 (only SIP <=> SIP) | Fail | PlusNet didn´t support G.722 |
4.3.7 | Normal call | Test call with codec G.729 | Pass |
|
4.3.2.8 | Clip No Screening | Outgoing call from PBX to PSTN - With feature Clip No Screening
- Test with several different A-numbers
- If two or more locations with different area codes are assigned to one SIP trunk, the number in the PAI header has to be the geographic number which belongs to the users location. This is very important in case of emergency calls, because PlusNet uses the number
| Pass |
|
4.3.2.8.1 | Clip No Screening | Despite Clip No Screening the geographic number in the PAI header corresponds to the location of the user | Pass | Is it possible to configure several PAI per SIP trunk? Yes |
4.3.2.8.2 | Clip No Screening | Display of A-number (NoSClip) at B-party (PSTN) | Pass |
|
4.3.2.8.3 | Clip No Screening | Display of A-number (NoSClip) at B-party (international PSTN; depending on the destination carrier, the NoSClip telephone number may not be displayed in this case!) | Pass |
|
4.3.2.8.3 | Clip No Screening | Call made from a PSTN line to an IP-PBX line with call forward to a line within the same IP-PBX, Answer Call.- Either party terminates call.
| Pass | Does the IP-PBX has configuration settings to send SIP status 181 messages to the soft switch? Yes |
4.3.2.8.4 | Clip No Screening | Display of A-number (NoSClip) at B-party (mobile) | Pass |
|
4.3.3.9 | Special call situations | Outgoing call PBX => PSTN - Call is rejected by B-party
| Pass |
|
4.3.3.10 | Special call situations | Outgoing call PBX=> PSTN - B-party does not answer; clearing after timer
| Pass |
|
4.3.3.11 | Special call situations | Outgoing call PBX => PSTN | Pass |
|
4.3.3.12 | Special call situations | Outgoing call PBX=> PSTN - A-party hangs up before call is established (cancel)
| Pass |
|
4.3.3.13 | Special call situations | Incoming call PSTN => PBX - Call is rejected by PBX party
| Pass |
|
4.3.3.14 | Special call situations | Incoming call PSTN => PBX - PBX party does not answer; clearing after timer
| Pass |
|
4.3.3.15 | Special call situations | Incoming call PSTN => PBX - PBX party busy; busy tone
| Pass |
|
4.3.3.16 | Special call situations | Incoming call PSTN => PBX - A-party hangs up before call is established (cancel)
| Pass |
|
4.3.4.17 | Call clearing | Incoming / outgoing call; clearing after established call. Correct clearing on both sides - PBX party hangs up
- PSTN party hangs up
| Pass |
|
4.3.4.18 | Call clearing | Interrupting the network connection of the SIP terminal device during a call - Call should be cleared correctly
| Pass |
|
4.4.19 | Hold | PBX => PSTN and PSTN => PBX - Test call in both directions
| Pass |
|
4.4.19.1 | Hold | Putting an external call on hold in the PBX | Pass |
|
4.4.19.2 | Hold | If applicable, MoH (music on hold) at A-party (PSTN) | Pass |
|
4.4.19.3 | Hold | HOLD RETRIEVE: retrieving the external call | Pass |
|
4.4.19.4 | Hold | Clearing the connection of the A-party while it is put on hold | Pass |
|
4.4.20 | Hold | PBX => PSTN and PSTN => PBX - Test call in both directions
| Pass |
|
4.4.20.1 | Hold | Putting an external call on hold in the PSTN | Pass |
|
4.4.20.2 | Hold | If applicable, MoH (music on hold) at A-party (PBX) | Pass |
|
4.4.20.3 | Hold | HOLD RETRIEVE: retrieving the external call | Pass |
|
4.4.20.4 | Hold | Clearing the connection of the A-party while it is put on hold | Pass |
|
4.5.21 | Call transfer | Internal call is transferred to external party: internal => PBX => external | Pass |
|
4.5.21.1 | Call transfer | Call transfer from PBX party => PSTN party with announcement (attendant transfer) | Pass |
|
4.5.21.2 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer) | Pass |
|
4.5.21.3 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer) - Call is rejected or not answered
| Pass |
|
4.5.21.4 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer) | Pass |
|
4.5.22 | Call transfer | Transferred call from external party to PBX: PSTN => PBX | Pass |
|
4.5.22.1 | Call transfer | Call transfer from PSTN => PBX party with announcement (attendant transfer) | Pass |
|
4.5.22.2 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer) | Pass |
|
4.5.22.3 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer) - Call is rejected or not answered
| Pass |
|
4.5.22.4 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer) | Pass |
|
4.5.23 | Call transfer | Call from external party transferred to another external party: external => PBX => external | Pass |
|
4.5.23.1 | Call transfer | PSTN => PBX party => PSTN with announcement (attendant transfer) | Pass |
|
4.5.23.2 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer) | Pass |
|
4.5.23.3 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer) - Call is rejected or not answered
| Pass |
|
4.5.23.4 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer) | Pass |
|
4.6.24 | Call diversion | PBX party CFU to external party (PSTN) | Pass |
|
4.6.24.1 | Call diversion | Internal call (CFU) => PSTN | Pass |
|
4.6.24.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.24.3 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.24.4 | Call diversion | External => PBX party (CFU) => PSTN | Pass |
|
4.6.24.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.24.6 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.25 | Call diversion | PBX party CFNR to external party | Pass |
|
4.6.25.1 | Call diversion | Internal call (CFNR) => PSTN | Pass |
|
4.6.25.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.25.3 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.25.4 | Call diversion | External => PBX party (CFNR) => PSTN | Pass |
|
4.6.25.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.25.6 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.26 | Call diversion | PBX party CFB to external party | Pass |
|
4.6.26.1 | Call diversion | Internal call (CFB) => PSTN | Pass |
|
4.6.26.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.26.3 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.26.4 | Call diversion | External => PBX party (CFB) => PSTN | Pass |
|
4.6.26.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.26.6 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.27 | Call diversion | External party (PSTN, mobile, etc.) CFU to PBX party: | Pass |
|
4.6.27.1 | Call diversion | External (CFU) => PBX party | Pass |
|
4.6.27.2 | Call diversion | PBX party busy, rejects call, does not answer | Pass |
|
4.6.27.3 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.27.4 | Call diversion | External (CFNR) => PBX party | Pass |
|
4.6.27.5 | Call diversion | PBX party busy, rejects call, does not answer | Pass |
|
4.6.27.6 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.28 | Call diversion | External party (PSTN, mobile, etc.) CFB to PBX party: | Pass |
|
4.6.28.1 | Call diversion | PBX party busy, rejects call, does not answer | Pass |
|
4.6.28.2 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.29 | Call diversion | Call deflection: diversion during the ring phase | Pass |
|
4.6.29.1.1 | Call diversion | Internal call PBX party CD => PBX party | Pass |
|
4.6.29.1.2 | Call diversion | PBX party busy | Pass |
|
4.6.29.1.3 | Call diversion | PBX party does not answer | Pass |
|
4.6.29.1.4 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.29.2.1 | Call diversion | External call to PBX party CD => PBX party | Pass |
|
4.6.29.2.2 | Call diversion | PBX party busy | Pass |
|
4.6.29.2.3 | Call diversion | PBX party does not answer | Pass |
|
4.6.29.2.4 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.29.3.1 | Call diversion | External call to PBX party CD => external PSTN party | Pass |
|
4.6.29.3.2 | Call diversion | PSTN party busy | Pass |
|
4.6.29.3.3 | Call diversion | PSTN party does not answer | Pass |
|
4.6.29.3.4 | Call diversion | A-party clears in ring phase | Pass |
|
4.7.30 | Call waiting | Incoming call during active internal call | Pass |
|
4.7.30.1 | Call waiting | Call waiting tone | Pass |
|
4.7.30.2 | Call waiting | Display of number of waiting party | Pass |
|
4.7.30.3 | Call waiting | Acceptance of waiting call | Pass |
|
4.7.30.4 | Call waiting | Putting the waiting call on hold | Pass |
|
4.7.30.5 | Call waiting | Retrieve of waiting call | Pass |
|
4.7.30.6 | Call waiting | Holding the 2nd call | Pass |
|
4.7.30.7 | Call waiting | Terminating the active call | Pass |
|
4.7.30.8 | Call waiting | Disregarding the waiting call | Pass |
|
4.7.30.9 | Call waiting | Rejecting the waiting call | Pass |
|
4.8.31 | 3-party conference | Establishing a conference according to the operating instructions of the PBX Internal - internal - external | Pass |
|
4.8.31.1 | 3-party conference | Selecting a party (internal or external); 3rd party is put on hold - Switching to 3rd party; 2nd party is put on hold
- Reactivating the conference
- Clearing one party (internal or external)
- Terminating the conference
| Pass |
|
4.8.32 | 3-party conference | Establishing a conference according to the operating instructions of the PBX internal - external - external | Pass |
|
4.8.32.1 | 3-party conference | Selecting a party (external); 3rd party is put on hold - Switching to 3rd party; 2nd party is put on hold
- Reactivating the conference
- Clearing one party (external)
- Terminating the conference
| Pass |
|
4.9.33 | Pick up | Picking up a call from another extension of the PBX | Pass |
|
4.10.34 | Call list | Entries in the call list - Incoming from PSTN
- Incoming from mobile
- Incoming CLIR
- Dialing prefix for outside line in the call list
| Pass |
|
4.10.35 | Call list | Call-back from call list | Pass |
|
4.11.36 | DTMF | DTMF support (G.711) - PSTN => PBX (SIP terminal device)
- PSTN => PBX (analog or system terminal device)
- PBX (SIP terminal device) => PSTN
- PBX (analog or system terminal device) => PSTN
| Pass |
|
4.11.37 | DTMF | DTMF support (G.729) - PSTN => PBX (SIP terminal device) - PSTN => PBX (analog or system terminal device) - PBX (SIP terminal device) => PSTN - PBX (analog or system terminal device) => PSTN | Pass |
|
4.12.38 | Fax | Fax reception (G.711 only) | Pass |
|
4.12.38.1 | Fax | Network-side T.38 re-invite rejected by PBX (response 488) or only G.711 codec is confirmed - One-page fax
- Multi-page fax (at least 5 pages)
| Pass |
|
4.12.39 | Fax | Fax sending (G.711 only) | Pass |
|
4.12.39.1 | Fax | Network-side T.38 re-invite rejected by PBX (response 488) or only G.711 codec is confirmed - One-page fax
- Multi-page fax (at least 5 pages)
| Pass |
|
4.12.40 | Fax | Fax reception via T.38 | Pass |
|
4.12.40.1 | Fax | Re-invite to T.38 by PBX or network - One-page fax - Multi-page fax (at least 5 pages) | Pass |
|
4.12.41 | Fax | Fax sending via T.38 (not possible in conjunction with the encryption option) | Pass |
|
4.12.41.1 | Fax | Re-invite to T.38 by PBX or network - T.38-only invites are not supported
- One-page fax
- Multi-page fax (at least 5 pages)
| Pass |
|
5.42 | Redundancy | Test of redundancy: - Only if redundant connection is possible on the PBX side
- This requires at least two PBX servers to be online on the SIP trunk in registration mode, or exactly two PBX servers in peering mode.
| Pass |
|
5.42.1 | Redundancy | 1. Register all available PBX systems the Plusnet SBC. 2. Calls from Plusnet => PBX are routed by round robin procedure 3. Deliberately de-register one PBX system => no more calls are routed to this PBX 4. and/or disconnect the PBX system from LAN => after the register expire period, no more calls are routed to this PBX. 5. Calls are only routed to the remaining PBX systems. | Pass |
|
Conclusion
This Application Notes document describes the steps required to configure the Ribbon SBC 1000/2000 to successfully interoperate with the Cisco CUCM and PlusNet SIP Trunk. All feature and serviceability test cases have been completed. The majority of test cases passed with noted exceptions and observations provided in Interoperability Test Results.
Overview
This Application Note is a configuration guide for the Ribbon SBC (Session Border Controller) 1000/2000 when connecting to Cisco Unified Communication Manager (CUCM) and PlusNet SIP Trunk.
The configuration guide supports features outlined in the Microsoft Technet web page:
Introduction
Interoperability compliance testing focuses on verifying inbound and outbound call flows between Ribbon SBC 1000/2000 and Cisco CUCM.
Audience
This technical document provides telecommunications engineers with information for configuring both the Ribbon SBC and the third-party product. Procedures in this document require navigating third-party equipment as well as applying Ribbon SBC Command Line Interface (CLI) commands. To complete the configuration and perform any troubleshooting, the engineer performing the procedures must understand the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP.
Requirements
The following table lists the hardware and software used in the reference configuration.
Test Equipment and Software
Vendor | Equipment | Software Version |
---|
Ribbon Networks | SBC 2000 | V8.0.2 |
Third-party Vendor |
Cisco | Cisco Unified CM Administration | 12.0.1.21900 |
Cisco | Cisco SIP Phone 7841 | sip78xx.11-7-1-17 |
VentaFax | Fax Machine VentaFax | 7.6.243.616 |
Reference Configuration
The following figure serves as a topology for the reference configuration. The figure shows the connectivity between third-party equipment and the Ribbon SBC 1000/2000.
Reference Configuration Topology
Support
For questions about information in this document, contact Ribbon Support in either of the following ways:
Verify License
The interoperability test described in this document requires no special licensing.
1. Security Profile
Select System > Security > SIP Trunk Security Profile
SIP Trunk Security Profile
2. SIP Profile
Select Device > Device Settings > SIP Profile
3. SIP Trunk
Select Device > Trunk > Add New
4. Route Group
Select Call Routing > Route/Hunt > Route Group > Add New
5. Route List
Select Call Routing > Route/Hunt > Route List > Add New
6. Route Pattern
Select Call Routing > Route/Hunt > Route Pattern > Add New
Ribbon SBC 1000/2000 Configuration
The following configuration steps provide an example of how to configure the Ribbon SBC 1000/2000 to interoperate with Skype 2015 and Virgin Media SIP Trunk:
- SIP Profile
- SIP Server
- Media System
- Media Profiles
- Media List
- Remote Authorization Tables
- Signaling Groups
- Transformation
- Call Routing Table
1. SIP Profile
SIP Profiles control how the Ribbon SBC 1000/2000 communicates with SIP devices. The profiles control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags.
Select Settings > SIP > SIP Profiles to access the SIP Profile screen.
The following figures show the default SIP profile used for the Ribbon 1000/2000 used for this configuration effort.
2. SIP Server
SIP Server Tables contain information about the SIP devices connected to the Ribbon SBC 1000/2000.
Select Settings > SIP > SIP Server Tables to access the SIP Server Tables screen.
The entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with each SIP server. The entries also contain links to counters that are useful for troubleshooting, as shown in the following figures.
The Media System Configuration contains system-wide settings for the Media System. Configuring the media system means setting the number of RTP/RTCP port pairs and the starting port.
Select Settings > Media > Media System Configuration to access the Media System configuration screen.
Media Profiles specify the individual voice and fax compression codecs and their associated settings for inclusion into a Media List. Different codecs provide varying levels of compression, allowing the reduction of bandwidth requirements at the expense of voice quality.
Select Settings > Media > Media Profiles.
The following figures illustrate possible media profiles of the voice codecs used for the SBC 1000/2000. The examples are for reference only.
The Media List shows the selected voice and fax compression codecs and their associated settings.
Select Settings > Media > Media List to access the Media List configuration screen.
6. Remote Authorization Tables
Remote Authorization Tables and their entries contain information used to respond to request message challenges by an upstream server. The Remote Authorization Tables on this page appear as options in Creating and Modifying Entries in the SIP Servers (For additional information about Remote Authorization Tables, see the Ribbon online SBC 1000/2000 documentation).
Select Settings > SIP > Remote Authorization Tables to access the Remote Authorization Tables configuration screen.
Remote Authorization Table
7. Signaling Groups
Signaling Groups allow telephony channels to be grouped for routing and shared configuration. These groups are the entity to which calls are routed, and the location from which Call Routes are selected. These are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, Signaling Groups will specify protocol settings and links to server, media, and mapping tables.
Select Settings > Signaling Groups to access the Signaling Groups configuration screens.
Transformation Tables facilitate the conversion of names, numbers, and other fields when routing a call. For example, transformations can convert a public PSTN number into a private extension number, or a SIP address (URI). Every entry in a Call Routing Table requires a Transformation Table.
Select Settings > Transformation to access the Transformation configuration screen.
9. Call Routing Table
Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports, and between protocols (like ISDN to SIP). Call Routing Tables define routes. The use of Call Routing Tables allows for flexible configuration of which calls will be carried, and also how the calls are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message translations, Cause Code Reroutes, Media Lists, and the three types of Signaling Groups (ISDN, SIP and, CAS).
Select Settings > Call Routing Table to access the Call Routing Table configuration screen.
Interoperability Test Results
The following table provides test results for interoperability compliance testing between Ribbon SBC 1000/2000 and CUCM
Interoperability Compliance Test Results
Test Number | Test Scenario | Setup / Test Results | Status | Comment |
---|
4.1 | Registration and authentication (registration mode) | The PBX is able to execute the correct resolution oft he DNS SRV record | Pass |
|
4.3 | Basic call | With the basic call tests, the standard call scenarios and the CLIP/CLIR features are tested. - Only en-bloc dialing is supported, overlap sending is not possible.
| Pass |
|
4.3.1 | Normal call | Outgoing call from PBX to PSTN - En-bloc dialling
- Local area call (without area code); area code must be set by the PBX
- Setting of the correct calling number with all available telephone number blocks
- If two or more locations with different area codes are assigned to one SIP trunk, the number in the PAI header has to be the geographic number which belongs to the users location. This is very important in case of emergency calls, because Plusnet uses the number of the PAI header to route the emergency call to the proper emergency call center.
| Pass |
|
4.3.1.1 | Normal call | The geographic number in the PAI header corresponds to the location 11 of the user | Pass | Is it possible to configure several PAI per SIP trunk? Yes |
4.3.1.2 | Normal call | Display of A-number in B-party CLIP (national PSTN) | Pass |
|
4.3.1.3 | Normal call | Display of A-number in B-party CLIP (international PSTN) | Pass |
|
4.3.1.4 | Normal call | Display of A-number in B-party CLIP (mobile) | Pass |
|
4.3.1.5 | Normal call | Call to mobile Outgoing call to mobile => mobile phone turned off | Pass |
|
4.3.1.6 | Normal call | Suppression of A-number => CUR | Pass |
|
4.3.1.7 | Normal call | Outgoing call from analog extension | Pass |
|
4.3.1.8 | Normal call | Outgoing call (> 5 min.) => PSTN | Pass | Hold connection for 5 minutes => RTP still correct? Yes |
4.3.2 | Normal call | Incoming call from PSTN (national) => PBX - Test all available telephone number blocks
| Pass |
|
4.3.2.1 | Normal call | Display of A-number => CLIP | Pass |
|
4.3.2.2 | Normal call | Incoming call from mobile => PBX Display of A-number => CLIP | Pass |
|
4.3.2.3 | Normal call | Suppression of A-number => CLIR | Pass |
|
4.3.3 | Normal call | Two simultaneous outgoing/incoming calls | Pass |
|
4.3.4 | Normal call | Enabled feature DND (do not disturb) | Pass |
|
4.3.5 | Normal call | Test call with codec G.711 | Pass |
|
4.3.6 | Normal call | Test call with codec G.722 (only SIP <=> SIP) | Fail | PlusNet didn´t support G.722 |
4.3.7 | Normal call | Test call with codec G.729 | Pass |
|
4.3.2.8 | Clip No Screening | Outgoing call from PBX to PSTN - With feature Clip No Screening
- Test with several different A-numbers
- If two or more locations with different area codes are assigned to one SIP trunk, the number in the PAI header has to be the geographic number which belongs to the users location. This is very important in case of emergency calls, because PlusNet uses the number
| Pass |
|
4.3.2.8.1 | Clip No Screening | Despite Clip No Screening the geographic number in the PAI header corresponds to the location of the user | Pass | Is it possible to configure several PAI per SIP trunk? Yes |
4.3.2.8.2 | Clip No Screening | Display of A-number (NoSClip) at B-party (PSTN) | Pass |
|
4.3.2.8.3 | Clip No Screening | Display of A-number (NoSClip) at B-party (international PSTN; depending on the destination carrier, the NoSClip telephone number may not be displayed in this case!) | Pass |
|
4.3.2.8.3 | Clip No Screening | Call made from a PSTN line to an IP-PBX line with call forward to a line within the same IP-PBX, Answer Call.- Either party terminates call.
| Pass | Does the IP-PBX has configuration settings to send SIP status 181 messages to the soft switch? Yes |
4.3.2.8.4 | Clip No Screening | Display of A-number (NoSClip) at B-party (mobile) | Pass |
|
4.3.3.9 | Special call situations | Outgoing call PBX => PSTN - Call is rejected by B-party
| Pass |
|
4.3.3.10 | Special call situations | Outgoing call PBX=> PSTN - B-party does not answer; clearing after timer
| Pass |
|
4.3.3.11 | Special call situations | Outgoing call PBX => PSTN | Pass |
|
4.3.3.12 | Special call situations | Outgoing call PBX=> PSTN - A-party hangs up before call is established (cancel)
| Pass |
|
4.3.3.13 | Special call situations | Incoming call PSTN => PBX - Call is rejected by PBX party
| Pass |
|
4.3.3.14 | Special call situations | Incoming call PSTN => PBX - PBX party does not answer; clearing after timer
| Pass |
|
4.3.3.15 | Special call situations | Incoming call PSTN => PBX - PBX party busy; busy tone
| Pass |
|
4.3.3.16 | Special call situations | Incoming call PSTN => PBX - A-party hangs up before call is established (cancel)
| Pass |
|
4.3.4.17 | Call clearing | Incoming / outgoing call; clearing after established call. Correct clearing on both sides - PBX party hangs up
- PSTN party hangs up
| Pass |
|
4.3.4.18 | Call clearing | Interrupting the network connection of the SIP terminal device during a call - Call should be cleared correctly
| Pass |
|
4.4.19 | Hold | PBX => PSTN and PSTN => PBX - Test call in both directions
| Pass |
|
4.4.19.1 | Hold | Putting an external call on hold in the PBX | Pass |
|
4.4.19.2 | Hold | If applicable, MoH (music on hold) at A-party (PSTN) | Pass |
|
4.4.19.3 | Hold | HOLD RETRIEVE: retrieving the external call | Pass |
|
4.4.19.4 | Hold | Clearing the connection of the A-party while it is put on hold | Pass |
|
4.4.20 | Hold | PBX => PSTN and PSTN => PBX - Test call in both directions
| Pass |
|
4.4.20.1 | Hold | Putting an external call on hold in the PSTN | Pass |
|
4.4.20.2 | Hold | If applicable, MoH (music on hold) at A-party (PBX) | Pass |
|
4.4.20.3 | Hold | HOLD RETRIEVE: retrieving the external call | Pass |
|
4.4.20.4 | Hold | Clearing the connection of the A-party while it is put on hold | Pass |
|
4.5.21 | Call transfer | Internal call is transferred to external party: internal => PBX => external | Pass |
|
4.5.21.1 | Call transfer | Call transfer from PBX party => PSTN party with announcement (attendant transfer) | Pass |
|
4.5.21.2 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer) | Pass |
|
4.5.21.3 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer) - Call is rejected or not answered
| Pass |
|
4.5.21.4 | Call transfer | Call transfer from PBX party => PSTN party without announcement (blind transfer) | Pass |
|
4.5.22 | Call transfer | Transferred call from external party to PBX: PSTN => PBX | Pass |
|
4.5.22.1 | Call transfer | Call transfer from PSTN => PBX party with announcement (attendant transfer) | Pass |
|
4.5.22.2 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer) | Pass |
|
4.5.22.3 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer) - Call is rejected or not answered
| Pass |
|
4.5.22.4 | Call transfer | Call transfer from PSTN => PBX party without announcement (blind transfer) | Pass |
|
4.5.23 | Call transfer | Call from external party transferred to another external party: external => PBX => external | Pass |
|
4.5.23.1 | Call transfer | PSTN => PBX party => PSTN with announcement (attendant transfer) | Pass |
|
4.5.23.2 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer) | Pass |
|
4.5.23.3 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer) - Call is rejected or not answered
| Pass |
|
4.5.23.4 | Call transfer | PSTN => PBX party => PSTN without announcement (blind transfer) | Pass |
|
4.6.24 | Call diversion | PBX party CFU to external party (PSTN) | Pass |
|
4.6.24.1 | Call diversion | Internal call (CFU) => PSTN | Pass |
|
4.6.24.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.24.3 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.24.4 | Call diversion | External => PBX party (CFU) => PSTN | Pass |
|
4.6.24.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.24.6 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.25 | Call diversion | PBX party CFNR to external party | Pass |
|
4.6.25.1 | Call diversion | Internal call (CFNR) => PSTN | Pass |
|
4.6.25.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.25.3 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.25.4 | Call diversion | External => PBX party (CFNR) => PSTN | Pass |
|
4.6.25.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.25.6 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.26 | Call diversion | PBX party CFB to external party | Pass |
|
4.6.26.1 | Call diversion | Internal call (CFB) => PSTN | Pass |
|
4.6.26.2 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.26.3 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.26.4 | Call diversion | External => PBX party (CFB) => PSTN | Pass |
|
4.6.26.5 | Call diversion | PSTN party busy, rejects call, does not answer | Pass |
|
4.6.26.6 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.27 | Call diversion | External party (PSTN, mobile, etc.) CFU to PBX party: | Pass |
|
4.6.27.1 | Call diversion | External (CFU) => PBX party | Pass |
|
4.6.27.2 | Call diversion | PBX party busy, rejects call, does not answer | Pass |
|
4.6.27.3 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.27.4 | Call diversion | External (CFNR) => PBX party | Pass |
|
4.6.27.5 | Call diversion | PBX party busy, rejects call, does not answer | Pass |
|
4.6.27.6 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.28 | Call diversion | External party (PSTN, mobile, etc.) CFB to PBX party: | Pass |
|
4.6.28.1 | Call diversion | PBX party busy, rejects call, does not answer | Pass |
|
4.6.28.2 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.29 | Call diversion | Call deflection: diversion during the ring phase | Pass |
|
4.6.29.1.1 | Call diversion | Internal call PBX party CD => PBX party | Pass |
|
4.6.29.1.2 | Call diversion | PBX party busy | Pass |
|
4.6.29.1.3 | Call diversion | PBX party does not answer | Pass |
|
4.6.29.1.4 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.29.2.1 | Call diversion | External call to PBX party CD => PBX party | Pass |
|
4.6.29.2.2 | Call diversion | PBX party busy | Pass |
|
4.6.29.2.3 | Call diversion | PBX party does not answer | Pass |
|
4.6.29.2.4 | Call diversion | A-party clears in ring phase | Pass |
|
4.6.29.3.1 | Call diversion | External call to PBX party CD => external PSTN party | Pass |
|
4.6.29.3.2 | Call diversion | PSTN party busy | Pass |
|
4.6.29.3.3 | Call diversion | PSTN party does not answer | Pass |
|
4.6.29.3.4 | Call diversion | A-party clears in ring phase | Pass |
|
4.7.30 | Call waiting | Incoming call during active internal call | Pass |
|
4.7.30.1 | Call waiting | Call waiting tone | Pass |
|
4.7.30.2 | Call waiting | Display of number of waiting party | Pass |
|
4.7.30.3 | Call waiting | Acceptance of waiting call | Pass |
|
4.7.30.4 | Call waiting | Putting the waiting call on hold | Pass |
|
4.7.30.5 | Call waiting | Retrieve of waiting call | Pass |
|
4.7.30.6 | Call waiting | Holding the 2nd call | Pass |
|
4.7.30.7 | Call waiting | Terminating the active call | Pass |
|
4.7.30.8 | Call waiting | Disregarding the waiting call | Pass |
|
4.7.30.9 | Call waiting | Rejecting the waiting call | Pass |
|
4.8.31 | 3-party conference | Establishing a conference according to the operating instructions of the PBX Internal - internal - external | Pass |
|
4.8.31.1 | 3-party conference | Selecting a party (internal or external); 3rd party is put on hold - Switching to 3rd party; 2nd party is put on hold
- Reactivating the conference
- Clearing one party (internal or external)
- Terminating the conference
| Pass |
|
4.8.32 | 3-party conference | Establishing a conference according to the operating instructions of the PBX internal - external - external | Pass |
|
4.8.32.1 | 3-party conference | Selecting a party (external); 3rd party is put on hold - Switching to 3rd party; 2nd party is put on hold
- Reactivating the conference
- Clearing one party (external)
- Terminating the conference
| Pass |
|
4.9.33 | Pick up | Picking up a call from another extension of the PBX | Pass |
|
4.10.34 | Call list | Entries in the call list - Incoming from PSTN
- Incoming from mobile
- Incoming CLIR
- Dialing prefix for outside line in the call list
| Pass |
|
4.10.35 | Call list | Call-back from call list | Pass |
|
4.11.36 | DTMF | DTMF support (G.711) - PSTN => PBX (SIP terminal device)
- PSTN => PBX (analog or system terminal device)
- PBX (SIP terminal device) => PSTN
- PBX (analog or system terminal device) => PSTN
| Pass |
|
4.11.37 | DTMF | DTMF support (G.729) - PSTN => PBX (SIP terminal device) - PSTN => PBX (analog or system terminal device) - PBX (SIP terminal device) => PSTN - PBX (analog or system terminal device) => PSTN | Pass |
|
4.12.38 | Fax | Fax reception (G.711 only) | Pass |
|
4.12.38.1 | Fax | Network-side T.38 re-invite rejected by PBX (response 488) or only G.711 codec is confirmed - One-page fax
- Multi-page fax (at least 5 pages)
| Pass |
|
4.12.39 | Fax | Fax sending (G.711 only) | Pass |
|
4.12.39.1 | Fax | Network-side T.38 re-invite rejected by PBX (response 488) or only G.711 codec is confirmed - One-page fax
- Multi-page fax (at least 5 pages)
| Pass |
|
4.12.40 | Fax | Fax reception via T.38 | Pass |
|
4.12.40.1 | Fax | Re-invite to T.38 by PBX or network - One-page fax - Multi-page fax (at least 5 pages) | Pass |
|
4.12.41 | Fax | Fax sending via T.38 (not possible in conjunction with the encryption option) | Pass |
|
4.12.41.1 | Fax | Re-invite to T.38 by PBX or network - T.38-only invites are not supported
- One-page fax
- Multi-page fax (at least 5 pages)
| Pass |
|
5.42 | Redundancy | Test of redundancy: - Only if redundant connection is possible on the PBX side
- This requires at least two PBX servers to be online on the SIP trunk in registration mode, or exactly two PBX servers in peering mode.
| Pass |
|
5.42.1 | Redundancy | 1. Register all available PBX systems the Plusnet SBC. 2. Calls from Plusnet => PBX are routed by round robin procedure 3. Deliberately de-register one PBX system => no more calls are routed to this PBX 4. and/or disconnect the PBX system from LAN => after the register expire period, no more calls are routed to this PBX. 5. Calls are only routed to the remaining PBX systems. | Pass |
|
Conclusion
This Application Notes document describes the steps required to configure the Ribbon SBC 1000/2000 to successfully interoperate with the Cisco CUCM and PlusNet SIP Trunk. All feature and serviceability test cases have been completed. The majority of test cases passed with noted exceptions and observations provided in Interoperability Test Results.