Howto:DE - Deutsche Telefon - SIP Tk Anlagenanschluss SIP-Provider (2016)

Summary

The tests for the SIP Tk-Anlagenanschluss of the provider Deutsche Telefon were completed.

Issues found were:

SRTP

NAT Traversal

Redundancy

Correct signalling of Ringing-state

For more details about this issues, see the respective test-results sections.

Current test state

The tests for this product have been completed. See the Summary section for more details.

Testing of this product has been finalized June 10th, 2016.

Tests with MediaRelay

Registration

The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted. Moreover it requires all involved network elements to support IP-fragmentation.

CLIP

OK

CLIR

OK

Clip No Screening(CLNS)

OK

Additionally, redirection using "302 Moved Temporary" is possible. In case of a call forward, the incoming call is redirected to the PSTN. From point of view of the PBX the call doesn't exist after the redirection and is handled by the SIP-Provider. This is equivalent to "Partial Rerouting/Call Deflection" in ISDN. When a redirection is done, the redirection target will receive the original calling number, so for forwarded calls this will have the same effect as CLNS. However this redirection is not usable for Mobility calls or plain CLNS-calls. A plain CLNS-call is an outbound call with a "wrong" CGPN number (i.e. one that does not belong to the trunk). This happens e.g. if a call-center wants to send a service number (such 0800xxx) as CLI.

Codecs

The provider support the following codecs: G711A

Fax

Transport of faxes to/from the PSTN via G.711(A/U) codec was tested successfully. Additionally transport of faxes to/from the PSTN using the T.38 protocol was tested successfully. This is important for the innovaphone Fax-server. Even if the provider supports T.38, it is not guaranteed that all Fax-calls use T.38. However each call using T.38 will save you 2 DSP-licenses on the gateway hosting the Fax-interface.

SRTP

The provider does not support audio encryption using SRTP.

DTMF (RFC2833)

OK

NAT Traversal

The provider detects clients behind NAT and can handle calls to them without requiring the clients to use NAT-traversal methods like STUN. However MediaRelay is required, since the provider expects that all RTP-packets are passed through the PBX.

Reverse Media Negotiation

OK

Mobility Call

Transmitting DTMF-tones as SIP-INFO messages was not possible, however we can do Mobility-calls since we use MediaRelay on the SIP-Interface.

Redundancy

Registration of two SIP-interfaces on the same SIP-account is supported by the provider. However, the provider has no failover mechanism if one device is down. As a result, you can use both SIP-interfaces for load-balancing purposes. If one device is down, for a duration of up to 2 minutes (i.e registration interval) incoming and outgoing calls might be rejected/fail.

IP-Fragmentation

OK

Large SIP messages

OK

Correct signalling of Ringing-state

Ringing is not signalled by the provider. This will lead to incorrect call-state display on the PBX (phone-UI, myPBX, Soap) for outbound calls to the PSTN. The caller will see no status-update on the phone-display/PC-screen, showing that the remote party was reached and is ringing.
Additionally external callers forwarded/transferred back to the PSTN, will get no ring-tone but hear silence while the remote party is ringing. This silence while waiting might lead to aborting the call.

Early-Media

The provider supports early-media for outbound calls to the PSTN.

Session Timer

The tests regarding the SIP-session timer were successful. This means that either no session expiry is used or that it is used and works. It does not imply that session expiry actually is used.

Call Transfer

OK

Tests without MediaRelay

The tests without MediaRelay were aborted, since it is required by the provider. The reason for it, are audio problems when two external RTP-endpoints are connected(e.g. external transfer, mobility call).

Configuration

• Use profile DE-Deutsche Telefon-SIP Tk_Anlagenanschluss in the Gateway/Interfaces/SIP menu.

Contact

contact form of provider