> For the complete documentation index, see [llms.txt](https://prosbcdocs.telcobridges.com/llms.txt). Markdown versions of documentation pages are available by appending `.md` to page URLs; this page is available as [Markdown](https://prosbcdocs.telcobridges.com/configuration-details/configuration-by-use-case/rtp-no-anchoring/parameter-allow-low-delay-media-relay.md).

# Parameter: Allow low-delay media relay

#### What is "low-delay media relay"?

The low-delay media relay mode eliminates any VoIP audio delay caused by the SBC for IP to IP calls, by eliminating the need for jitter buffer, de-packetization and re-packetization.

With low-delay mode, packets get routed directly from one leg to the other with almost no delay (only Ethernet transit delay, generally a fraction of millisecond)

#### Which TelcoBridges products support this mode?

This mode is only available on SBC units (not on TMG units).

#### When is low-delay media relay effectively used?

Low-delay media-relay will be effective on a call only if:

* Both call legs have low-delay media relay option enabled
* Both call legs use [RTP port ranges](/configuration-details/configuration-by-use-case/rtp-no-anchoring/parameter-allow-low-delay-media-relay/configuring-an-ip-port-range.md) on SBC-protected VLANs (port range is configured per [NAP](/appendices/appendix-a-glossary/glossary-nap.md))
* Both call legs have been able to negotiate a compatible audio codec
* No IVR is required (play file, record file, play tone or detect tone)

Most calls going through a TelcoBridges SBC will use low-delay mode because:

* SIP codecs negotiation attempts to keep only compatible codecs by default
* No playback, recording, tone generation or tone detection is required by default

#### Will low-delay media relay affect SIP negociation?

Yes, enabling low-delay media relay will change how TelcoBridges SBC will perform codecs (SDP) negotiation between the two call legs.

When low-delay media relay is enabled, the SBC will invite the outgoing call leg with only codecs supported y incoming leg's SIP invite, in order to make sure that only compatible codecs are used.

If no compatible codecs can be found, the call will automatically use "transcoding" mode where all codecs from the profile will be allowed.

#### Can a call switch between low-delay and transcoding mode?

TelcoBridges SBC is able to seamlessly switch, during a call, from "low-delay" to "transcoding" mode, if conditions change during the call (re-invite causing incompatible audio codecs, or play/record/tone is required).

#### Any related options?

Enabling the low-delay mode will also allow a variant: Use the RTP no-anchoring mode.


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