Howto:DE - equada - VoIP Trunk SIP-Provider (2016)

Summary

PRELIMINARY REPORT -- Tests for this product are still ongoing (last updated August 29th, 2016) and may (and probably will) change.

Remarks

This provider enforces use of SRTP.

<internal>Provider SBC: TELES-SBC</internal>

List of Issues found in media-relay Configuration

FAX AUDIO

The provider does not fully support Audiofax (i.e. non-T.38)

REDIR 302

The provider does not support external call redirection using the SIP 302 Redirect response

REVERSE MEDIA

The provider does not support reverse media negotiation (a.k.a. late SDP)

SIP INFO

The provider does not support conveying DTMF using the SIP-INFO method.

XFER CONS EXT

The provider does not fully support external consultation call transfer scenarios.

XFER CONS

The provider does not fully support consultation call transfer after connect scenarios.

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, DTMF, EARLY_MEDIA_INBOUND, FAX_AUDIO, FAX_T38, 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, SIP_INFO, SRTP_INCOMING, SRTP_INTERNAL, SRTP_OUTGOING, 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 UDP, TCP and TLS as transport protocol. The tests were completed using TLS, since it offers encryption of the transmitted SIP-packets.

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 not successful. This will result in unwanted call termination on calls exceeding a certain time (default 30 minutes). Because of this, further tests were aborted.

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

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: or Diversion: SIP headers for providing the correct calling line id for diverted calls.

Early-Media

The provider supports early-media for outbound calls (hence inbound early media) to the PSTN.

Fax

Audio-Fax calls (that is, fax calls without T.38) do not work.

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 G711U

supported onnet (VoIP to VoIP): G711A, G711U, G722 and G729

IP-Fragmentation

OK

Large SIP messages

OK

Reverse Media Negotiation

Reverse-media negotiation is not supported. Therefore, a media-relay with exclusive coder configuration will be activated on the SIP-interface.

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 supports audio encryption using SRTP for incoming, outgoing and on-net calls.

Call Transfer

The provider does not handle internally transferred-after-connect calls.

The provider does not handle externally transferred calls.

Configuration

Use profile DE-equada-VoIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.

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.