# Configure ClearMode

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.

![](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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