Howto:Deutsche Telekom Magenta Zuhause SIP Provider Compatibility Test

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Innovaphone Compatibility Test Report

Contents

Summary

SIP Provider: Deutsche Telekom

The provider offers a product called Magenta Zuhause. It's a pure IP-based internet and telephony product, being connected via a DSL-modem to the Deutsche Telekom AG IP network.

The product is offered with up to 10 MSNs, from which each one has to be registered separately for use in the innovaphone gateway. Max. 2 concurrent calls are possible.

The provider offers some setup hints on his homepage.

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

That being said, the provider has achieved 77,5% of all possible test points (124/160). For more information on the test rating, please refer to the SIP Interop Test Description.

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 January 20th, 2015.

Testing Enviroment

Image:SIP_Compatibility_Testsetup.png

This scenario describes a setup where the PBX and phones are in a private network. The SIP trunk is configured with Media Relay and exclusive coder.

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 NOK
call using g723 NOK
call using g729 NOK
call using g722 NOK
Overlapped sending NOK
early media channel OK
Fax using T.38 NOK
Reverse Media Negotiation NOK
CGPN can be suppressed OK
CLIP no screening NOK
Long time call possible(>30 min) OK
External Transfer OK
NAT Detection NOK, STUN Server Usage required
Redundancy partly OK
SIP over TCP NOK
Voice Quality OK? OK

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 OK
DTMF tones received correctly - WQ connect call 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

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

Blind Transfer

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

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 used for tests have been operated with firmware V11r1rc5+ (110867). Following screenshots might be subject to change in future firmware releases.

SIP - Trunk

Deutsche Telekom will send you a document including your access data.

As this is in German, following a mapping table which data to use for the configuration:

  • "Ihre Anschlusskennung" - to be used as AOR (replace 00 by +49). Enter the MSN you want to register here.
  • "Ihre Zugangsnummer" - to be used as Username
  • "Ihr persönliches Kennwort" - to be used as password (insert twice)

Image:Telekom_SIP_Provider_Compatibility_Test_1.png

Number Mapping

Image:Telekom_SIP_Provider_Compatibility_Test_2.png

Route Settings

Image:Telekom_SIP_Provider_Compatibility_Test_3.png

  • Force Enblock is required since overlap dialing is not supported.
  • PPI Information must go in International format.

Note: In newer v11r2 versions the flag "NO ICE" should be set also.

Known Problems / Remarks

  • Redundancy: As each MSN has to be registered separately, routing for outgoing calls could be performed internally to allow bypassing failed registrations.

As this only works for outgoing calls, this feature is only rated with half.

  • NAT Detection: A STUN Server has to be configured according to the values recommended by the provider. STUN only works within a Full cone NAT, Restricted Cone or Port-Restricted Cone NAT-routed network. There's a way to investigate the NAT-Type used by your router.
  • The SIP provider rejects calls which try to establish an encrypted audio-stream, more precisely calls that have a crypto - attribute in their SDP. In v11 No encryption can be configured in the SIP-Interface menu, as shown above. In v10 this option does not exist. To configure No encryption in v10(or v9), you have to configure the SRTP dropdown-menu with an empty value.
  • The provider does not support CLNS (Clip No Screening) for this product
  • customers reported a problem where calls to certain numbers are rejected with cause code 415. According to a customer, configuring the SIP-Interface with "No-ICE" and Framesize of 20ms helped.
  • according to feedback from customers, this SIP-Trunk product of the provider does not support Call Deflection/redirection (i.e. same as Partial Rerouting in ISDN) using 302 Moved Temporary. If this feature is mandatory, you can use the DeutschlandLAN SIP-Trunk of Dt. Telekom or another SIP-provider that support 302 Moved Temporary (see Test-Reports section Clip No Screening (CLNS))

DeutschlandLAN IP Voice/Data S

We have not tested the DeutschlandLAN IP Voice/Data S product. However, we have had customer report which suggest that it works just the same. However, it seems that you need to use your MSN as AOR, instead of Ihre Anschlusskennung.

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