Howto:UPC Cablecom Provider Compatibility Test

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Innovaphone Compatibility Test Report

Contents

Summary

SIP Provider: UPC Cablecom

The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.

That being said, the provider has achieved 91% of all possible test points (143/157). For more information on the test rating, please refer to Test Description

The provider requires a SIP trunk without registration, for that reason this product is not suitable for innovaphone`s Hosting Scenario.

If the costumer doesn't have VPN contract, then the SIP Trunk it's done directly via Public IP address, so it's necessary to have the PBX/Media Gateway with Public IP address (this was the scenario tested).

If the customer has a UPC Cablecom VPN contract, direct IP routing between the SIP provider, the IP PBX and the IP phones is done. In this case, also QoS for the VoIP calls in the Cablecom network is used.

By default fax is possible only by G711 passthrough, however T.38 can be enabled at the provider for your specific trunk.

  • The sip trunk from upc cablecom supports the following features:
    • TCP oder UDP (according to prior agreement with upc cablecom)
    • SIP, RFC 2543 und 3261
    • SIP Privacy: id, RFC 3323
    • P-Asserted-Identity/Remote-Party-ID
    • Diversion Header
    • SIP OPTIONS (keep-alive)
    • DTMF RFC 2833
    • Codec
      • 1. Prio G.711a-law (with a frame length set at 20ms)
      • 2. Prio G.711u-law (with a frame length set at 20ms)
      • 3. G729
      • 4. G723
    • Fax: G.711a pass-through
    • Fax: T.38 (it's optional must be enabled)
  • The following features are not supported:
    • SIP Refer
    • SIP Update
    • SIP Info
    • SIP P-Preferred Identity
    • TLS/SRTP

Current test state

Image:Recprod.PNG The tests for this product have been completed and it has been approved as a recommended product (Certification document).

Testing of this product has been finalized June 18th, 2014.

Testing Enviroment

Image:SIPProviderTestTopology2.PNG

This scenario describes a setup where the PBX and phones are in a private network.

  • the SIP trunk is configured with Media Relay but without exclusive coder. This is the case when the test for "NAT Traversal" fails.

Test Results

For more information on the test procedure, please read the following wiki article: SIP Interop Test Description. Bold lines in the test results indicate a KO-criteria.

Basic Call

Tested feature Result
call using g711a OK
call using g711u OK
call using g723 OK
call using g729 OK
call using g722 NOK
Overlapped sending NOK
early media channel OK
Fax using T.38 OK
T.38 Transcoding by the provider NT*
Reverse Media Negotiation OK
CGPN can be suppressed OK
CLIP no screening OK
Long time call possible(>30 min) OK
External Transfer OK
NAT Detection NOK
Redundancy OK
SIP over TCP NOK
Voice Quality OK? OK
  • Since Media-Relay is being used no T.38 Transcoding test it's possible to be done with Analogue Endpoint on the Private Network.

Direct Dial In

Tested feature Result
Inbound(Provider -> Innovaphone) OK
Outbound(Innovaphone -> Provider) OK
Loop In call(Innovaphone -> Provider -> Innovaphone) OK

DTMF

Tested feature Result
DTMF tones sent correctly OK
DTMF tones sent correctly via SIP-Info NOK
DTMF tones received correctly OK

Hold/Retrieve

Tested feature Result
Call can be put on hold OK
Held end hears music on hold / announcement from PBX OK

Transfer with consultation

Tested feature Result
Call can be transferred OK
Held end hears music on hold OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok? MoH Ok?
inno1 calls inno2. inno2 transfers to PSTN-phone. OK OK
inno1 calls PSTN-phone. inno1 transfers to inno2. OK OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK OK
PSTN-phone calls inno1. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. OK OK

Transfer with consultation (alerting only)

Tested feature Result
Call can be transferred OK
Held end hears music on hold or dialling tone OK
Call returns to transferring device if the third

Endpoint is not available

OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok? MoH Ok?
inno1 calls inno2. inno2 transfers to PSTN-phone. OK OK
inno1 calls PSTN-phone. inno1 transfers to inno2. OK OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK OK
PSTN-phone calls inno1. PSTN-phone transfers to inno2. OK OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. OK OK

Blind Transfer

Tested feature Result
Call can be transferred OK
Held end hears dialling tone OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok?
inno1 calls inno2. inno2 transfers to PSTN-phone. OK
inno1 calls PSTN-phone. inno1 transfers to inno2. OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK
PSTN-phone calls inno1. PSTN-phone transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. OK

CFU / CFB Transfer

Tested feature Result
Call can be forward OK
Held end hears dialling tone OK

CFNR / Blind Transfer (alerting only)

Tested feature Result
Call can be transferred or forward OK
Held end hears dialling tone OK

The following tests are made to test if call transfer is working.

Tested feature Voice Ok?
inno1 calls inno2. inno2 transfers to PSTN-phone. OK
inno1 calls PSTN-phone. PSTN-phone transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to inno2. OK
PSTN-phone calls inno1. inno1 transfers to other PSTN-phone-2. OK

Broadcast Group & Waiting Queue

Tested feature Result
Caller can make a call to a Broadcast Group OK
Caller can make a call to a Waiting Queue OK
Announcement if nobody picks up the call OK

Configuration

Firmware version

All innovaphone devices use V10 sr10 firmware build.

SIP - Trunk

GW1 Interface to PBX

Image:UPC_Cablecom_-_SIP_Provider_Compatibility_Test_1a.png

GW2 Interface to SIP Provider

Image:UPC_Cablecom_-_SIP_Provider_Compatibility_Test_1b.png

Number Mapping

Image:UPC_Cablecom_-_SIP_Provider_Compatibility_Test_2.png

Route Settings

Image:UPC_Cablecom_-_SIP_Provider_Compatibility_Test_3.png

Redundancy

The UPC Cablecom can provide as redundancy mechanism:

  • Round robin and Backup functions (over SIP Option Ping) to trunk multiple PBX (Master and Standby).
  • Backup with more than 1 PBX or with call forwarding to a Mobile phone

Known Problems

  • In order for CLIP No Screening to work, we must configure the use of the P-Asserted Identity as shown below. It's also necessary that UPC Cablecom activates the Clip SA service on the SIP trunk.
http://x.x.x.x/!config add SIP /pai
http://x.x.x.x/!config write
http://x.x.x.x/!config activate
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