Howto:USA - Telnyx - SIP Trunk SIP-Provider (2025)

Summary

Tests for the SIP_Trunk SIP trunk product of the provider Telnyx were completed. Test results have been last updated on November 6th, 2025. Check the history of this article for the date of the first publication of the testreport.

Remarks

Telnyx offers a location SIP Server option for better connectivity in Europe with address sip.telnyx.eu, this is the default SIP Server configured by the Profile, if you are located in America you can manually change to sip.telnyx.com while inserting the account details.

List of Issues found in media-relay Configuration

CLNS

Outgoing calls cannot be sent with a foreign calling party number (CLI).

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 T38ANDAUDIO

The provider does not support fallback to audio-fax if T.38 fails.

MOBILITY

The provider can not send DTMF signals via SIP-INFO messages.

REDIR 302

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

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

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.

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 2 minutes incoming and outgoing calls might be rejected/fail.

Correct signalling of Ringing-state

OK

CLIR

OK

Clip No Screening (CLNS)

CLIP no screening (CLNS) is not possible, that is, outgoing calls can only have the own subscriber number (or extension), no foreign numbers as calling line id. This affects for example externally forwarded and mobility calls. However, on-net (that is, from SIP provider to another customer at the same SIP provider) CLIP no screening (CLNS) is possible. As this is a non-German provider, the issue with off-net CLNS could be related to the provider or related to the international PSTN peering. Please consult the SIP provider if CLNS will work for you.

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. However, all fax endpoints must be configured with exclusive codec "G711A".

Template:SIP Profile Test T38 PSTN yes onnet no fallback no

Codecs

supported to/from PSTN: G711A, G711U and G722

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 USA-Telnyx-SIP_Trunk in Gateway/Interfaces/SIP to configure this SIP provider.

Please note the following configuration hints:

• FAX requires exclusive G711A codec
• If you intend to use SIPS (SIP/TLS) registration, you need to add the ' USERTrust RSA Certification Authority' certificate to the trust list of your SBC
• Dialling of subscriber numbers not possible, 'Dialing Location' must be configured without 'Area Code'

A most recent v15r1 firmware is required to use this SIP-profile. For hints regarding upgrade to v15r1, see Howto15r1:Firmware_Upgrade_V14r2_V15r1

New profiles are added in the course of our V15R1 software Service Releases, see ReleaseNotes15r1:Firmware. Here is an up to date list of tested SIP providers.

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.