Howto:DE - EWE - business DSL voice plus SIP-Provider (2018)
Summary
PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated February 5th, 2018) and may (and probably will) change.
<internal>Provider SBC: ewetel/oldenburg-(prod)</internal>
List of Issues found in no media-relay Configuration
This is a list of all issues found in a configuration where the media stream between endpoints and the SIP provider - as opposed to the signalling - is not routed through the SBC.
- CONN NR INCOMING
- Incoming calls from the PSTN don't show a correct connected number to the calling party.
- MOBILITY
- The provider can not send DTMF signals via SIP-INFO messages.
- SDP VIDEO
- The provider does not support receiving video media capabilities in the SDP-part of a SIP message.
- SIP INFO
- The provider does not support conveying DTMF using the SIP-INFO method.
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
List of Issues found in media-relay Configuration
This section lists the results that differ from the results for the first configuration.
- MOBILITY
- This feature, which does not work in the first configuration, works fine in the second configuration.
Test Results
This section explains the test results for all possible configurations in more detail.
Configuration without media-relay
- Registration
- The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted(TLS). Moreover it 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.
- 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 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 600 seconds incoming and outgoing calls might be rejected/fail.
- Correct signalling of Ringing-state
- OK
- CLIR
- OK
- 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:
orDiversion:
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, this configuration is not strictly required, as the provider supports clip no screening so that redirected calls (i.e. call forwards to external numbers) will show a proper calling line id (CLI) at the receiving party anyway. However, it may be useful anyhow to get rid of externally forwarded calls on the SBC entirely.
- COLP
- Outbound calls to the PSTN show the correct connected number. However incoming calls from the PSTN do not.
- A caller from the PSTN will receive an incorrect connected number that differs from the dialled number. This might lead to the caller cancelling the call.
- For outbound calls to the PSTN, an update of the connected number is not signalled to the caller.
- Early-Media
- The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
- Fax
- Transport of faxes to/from the PSTN via G.711 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.
- Codecs
- supported to/from PSTN: G711A
- supported onnet (VoIP to VoIP): G711A
- IP-Fragmentation
- OK
- Large SIP messages
- OK
- Reverse Media Negotiation
- OK
- Mobility Calls
- Transmitting DTMF-tones as SIP-INFO messages is not supported. In a no-media-relay configuration, DTMF signalling can thus not be conveyed to the PBX. Mobility calls will not work.
- SRTP
- The provider does not support audio encryption using SRTP.
- Dialing of Subscriber Numbers
- OK
- Call Transfer
- OK
Configuration with media-relay
- Registration
- The provider supports only UDP as transport protocol. As a result, the SIP-communication is not encrypted(TLS). Moreover it 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.
- 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 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 600 seconds incoming and outgoing calls might be rejected/fail.
- Correct signalling of Ringing-state
- OK
- CLIR
- OK
- 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:
orDiversion:
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, this configuration is not strictly required, as the provider supports clip no screening so that redirected calls (i.e. call forwards to external numbers) will show a proper calling line id (CLI) at the receiving party anyway. However, it may be useful anyhow to get rid of externally forwarded calls on the SBC entirely.
- COLP
- Outbound calls to the PSTN show the correct connected number. However incoming calls from the PSTN do not.
- A caller from the PSTN will receive an incorrect connected number that differs from the dialled number. This might lead to the caller cancelling the call.
- For outbound calls to the PSTN, an update of the connected number is not signalled to the caller.
- Early-Media
- The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.
- Fax
- Transport of faxes to/from the PSTN via G.711 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.
- Codecs
- supported to/from PSTN: G711A and
- supported onnet (VoIP to VoIP): G711A
- 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
- OK
- Call Transfer
- OK
Configuration
The profile will appear in the list under Gateway/Interfaces/SIP when the tests are fully finished
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.