> 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/tmedia-documentation/legacy-releases-toolpack-v2.x/configure-clearmode.md).

# Configure ClearMode

#### ***Applies to version(s): v2.7 and above***

Clearmode is defined in RFC4040 and is fully supported and does not required any special licensing or configuration from the Web portal to enable it.

### For ISDN/SS7 to SIP calls

If the SETUP/IAM message have its Information Transfer Rate and Information Transfer Capability set to 64kbits/sec Unrestricted, the Codec will be force to Clear Channel on the SIP side.

This will result in:

```
m=audio 0 RTP/AVP 96

a=rtpmap:96 CLEARMODE/8000
```

By not specifying the ptime, this means it's 20ms.

If you want to use a packetization time different from 20ms, you have to go Profile->RTP and Audio->Clear Channel.

![profile clearmode](https://docs.telcobridges.com/w/images/c/c9/Profile_clearmode.png)

And this will result in a SDP like:

```
m=audio 0 RTP/AVP 96

a=rtpmap:96 CLEARMODE/8000

a=ptime:10
```

### For SIP to ISDN/SS7 calls

In the INVITE message, if the SDP is received with Clear Channel SDP (CLEARMODE/8000), the SETUP/IAM ISDN/SS7 Information Transfer Rate and Information Transfer Capability will be set accordingly (64kbits/sec Unrestricted).

Note: The use of Clear Channel is equivalent to the cost of G.711 codec density.

### For H.248 calls

The SDP in the profiles must be updated to support the CLEARMODE. Please go in **Profiles -> \[select Profile] -> SDP**.

Add **96** to the "m=audio" line

and add this line: `a=rtpmap:96 CLEARMODE/8000`

The final output will be something like this:

```
 m=audio 0 RTP/AVP 8 0 9 18 101 96
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15,32-36
 a=rtpmap:96 CLEARMODE/8000
```

### References

* [RFC 4040](http://tools.ietf.org/html/rfc4040)


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