Configure ClearMode
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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.
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
