Howto:Deutsche Telekom Magenta Zuhause SIP Provider Compatibility Test: Difference between revisions
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* 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 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 | * 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. | |||
[[Category:Compat|{{PAGENAME}}]] | [[Category:Compat|{{PAGENAME}}]] | ||
=== DeutschlandLAN IP Voice/Data S === | === 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''. | 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''. |
Revision as of 15:58, 3 May 2017
Innovaphone Compatibility Test Report
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
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
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)
Number Mapping
Route Settings
- 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.
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.