IP Call Control Module RTP Parameters

RTP codec list:
[rtp_codec] Specify the list of audio codecs to offer when originating a call. The valid codec names are 'pcmu' (PCM Mu-law) and 'pcma' (PCM A-law). Codec names should entered without quotes and be separated by a space. If left blank, a default list will be used.

Silence Control:
[rtp_silence_control] This parameter determines how silence in the outbound RTP stream is treated.

Advanced Settings

Frame Duration:
[rtp_frame_duration] This parameter configures the duration of outbound G.711 RTP frames in multiples of 10 milliseconds.

Jitter Buffer Depth:
[rtp_jitter_buffer_depth] This parameter sets the depth of the jitter buffer in multiples of 10 milliseconds.

T.38 offer as CED tone:
[t38_offer_as_ced] Specifies whether to generate a CED detected event when receiving a T.38 offer. A T.38 offer is a SIP re-Invite or H.323 requestMode message indicating an IP endpoint wishes to switch the IP call to T.38. This allows applications performing call progress to detect the T.38 offer and transition to fax.
false – Don’t send CED tone detected.
true – Send CED tone detected.

Type of Service (DSCP value):
[rtp_type_of_service] This parameter determines how the first six bits of the ToS field in the IP header are set for RTP packets.

Voice Frame Replacement:
[rtp_voice_frame_replacement] This parameter specifies how to treat missing inbound voice frames. Missing frames during G.711 fax calls will always be replaced with silence.