Howto:DE - Deutsche Telekom - CompanyFlex TCP SIP-Provider (2020)

From innovaphone wiki
Jump to navigation Jump to search

Summary

Tests for the CompanyFlex Complete TCP SIP trunk product of the provider Deutsche_Telekom were completed. Test results have been last updated on November 19th, 2020. Check the history of this article for the date of the first publication of the testreport. <internal>Provider SBC: BroadWorks</internal>

Note: For security reasons, Telekom limits all of your SIP Trunks, CompanyFlex connection allows several TCP sessions, but there are limits on messages per second based on a source IP address.

List of Issues found in media-relay Configuration

CLIR
The provider does not fully support suppression of the calling line id (CLIR) using the SIP Privacy: Id header.
EARLY MEDIA INBOUND
The provider does not support early-media (i.e. establish RTP-stream before 200 OK/connect) for calls to the PSTN.
FAX T38 ONNET
The provider does not support T.38 fax for onnet calls.
FAX T38
The provider does not fully support T.38 fax

Here is the list of test-cases that have been performed for this provider: 180_RINGING, BASIC_CALL, CLIR, CLNS_ONNET, CLNS, CONN_NR_DIFF, CONN_NR_INCOMING, CONN_NR, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38_ONNET, FAX_T38, FAX_T38ANDAUDIO, G711A_ONNET, G711A, G711U_ONNET, G711U, G722_ONNET, G722, G729_ONNET, G729, HOLD_RETRIEVE, IP_FRAGMENTATION, LARGE_SIP_MESSAGES, MOBILITY, OPUS_NB, OPUS_WB, RALERT_DISC, REDIR_302, REDIR_DIVHDR, REDIR_HISTHDR, REVERSE_MEDIA, SDP_ICE, SDP_RTCP_MUX, SDP_VIDEO, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, SUBSCRIBER_NR, XFER_BLIND, XFER_CONS_ALERT, XFER_CONS_EXT, XFER_CONS

Test Results

This SIP provider requires a media-relay configuration. That is, all media traffic between the SIP provider and all endpoints must flow through the SBC. For this reason, a configuration without media-relay has not been tested and hence no test results for this configuration are listed.

Configuration with media-relay

Registration
The provider supports only TCP as transport protocol. In general TCP is preferred to UDP, since UDP is an unreliable protocol and requires all involved network elements to support IP-fragmentation.
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 does not support a change of the remote RTP-endpoint during a call.
DTMF (RFC2833)
The provider can convey DTMF digits using the RTP payload method as per RFC2833.
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.
Redundancy
Registration of two SIP-interfaces on the same SIP-account is not supported by the provider. As a result, you cannot have a Standby gateway/PBX using the same account for failover or load-balancing purposes.
Correct signalling of Ringing-state
OK
CLIR
CLIR didn't work.
Clip No Screening (CLNS)
Straight clip no screening (i.e. sending a foreign number as calling line id) works fine. As this works anyway, it does not matter, if the provider supports the interpretation of History-Info: or Diversion: SIP headers for providing the correct calling line id for diverted calls. This provider supports call redirection using the SIP 302 Redirect header. Rerouting can be enabled by setting the Reroute supported check-mark in the corresponding Trunk object configuration. Also, the Interworking check-mark must be set in the route used for calls from and to the SIP provider.
However, during our test other interop problems were discovered when the Interworking Flag is enabled. Therefore it is not recommended to use the call redirection via SIP 302 Redirect header.
COLP
Outbound and inbound calls to/from the PSTN show the correct connected number.
For outbound calls to the PSTN, an update of the connected number is not signalled to the caller.
Early-Media
The provider does not support early (that is, before connect) media for outbound calls to the PSTN (hence no inbound early media). This may be an issue in cases where such media is played to the caller (e.g. when calling an unavailable mobile phone).
Fax
Transport of faxes to/from the PSTN via G.711 codec was tested successfully.
Transport of faxes using T.38 failed to PSTN and onnet destinations. Fallback to audio-fax worked.
As a result, T.38 is enabled on the SIP-interface, the use of audio-fax is necessary.
Codecs
supported to/from PSTN: G711A
supported onnet (VoIP to VoIP): G711A, G711U and G722
IP-Fragmentation
OK
Large SIP messages
OK
Reverse Media Negotiation
OK
Mobility Calls
Transmitting DTMF-tones as SIP-INFO messages is not supported, however mobility calls are still possible as in the media-relay configuration, the SBC will convey DTMF signalling to the PBX.
SRTP
The provider does not support audio encryption using SRTP.
Dialing of Subscriber Numbers
The provider does not support dialling numbers in subscriber number format. Make sure to configure the Dialing Location accordingly.
Call Transfer
OK


Configuration

Use profile DE-Deutsche_Telekom-CompanyFlex_TCP in Gateway/Interfaces/SIP to configure this SIP provider.

Please note the following configuration hints:

  • Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'
A most recent v12r2 firmware or higher is required to use this SIP-profile. For hints regarding upgrade to v12r2, see Howto:Firmware Upgrade V12r1 V12r2

New profiles are added in the course of our V12R2 software Service Releases, see Support:DVL-Roadmap Firmware 12r2. Here is an up to date list of tested SIP providers.

Multi Location

It's possible to have different MSNs and DDIs from different locations on the same trunk. But this is limited to 60 number objects. A number object is an MSN or the first DDI digit. So a complete DDI block from 0 to 9 (0-99 or 0-999) is counted as 10 number objects.

SIP Redundancy

In case of SIP redundancy the username should be in the format "+49199296000000XXXXXX@tel.t-online.de" so the feature "Call Routing for redundant SIP-Trunks" works correctly.

Disclaimer

These tests look at a number of interoperability scenarios between innovaphone SIP devices and a given SIP trunk product. As we are enhancing our testing procedures, nature and number of these tests will vary.

All test results document the fact how the tested combination performs in the tested scenario. It explicitly does not comment on the question what the reason is for the behaviour nor if and how it could be changed. It thus does not imply that either the SIP trunk provider or the innovaphone device fails in any way. It merely says that the combination does not perform as defined by the test.

If not mentioned otherwise, all tests are repeated on a weekly basis (a.k.a. nightly tests) using the then-current innovaphone firmware. If a tested combination keeps performing differently from the state documented here during nightly tests, we reserve the right to update this article accordingly.

Some tests do not have consistent results. This may occur for various reasons. If so, we document the tests result as being unstable.

Some of the tested SIP trunk products are not available for nightly tests. In this case, the fact will be noted in the Summary section.