CM: SIP SDP Telephone-event Codec Clock Rate Mismatch

Details

Communication Manager releases 6.x, 7.x, 8.0 and 8.0.1.

Problem Clarification

Telephone-event codec clock rate mismatch is seen in case of high bitrate (16000) DTMF codec in SDP offers.

Example: SDP offer to CM: v=0 o=BroadWorks 2442380 1 IN IP4 10.112.21.72 s=- c=IN IP4 10.112.21.72 t=0 0 m=audio 35278 RTP/AVP 111 108 8 101 96 a=rtpmap:111 AMR-WB/16000 a=fmtp:111 mode-set=0,1,2;mode-change-period=2;mode-change-capability=2 a=rtpmap:108 AMR/8000 a=fmtp:108 mode-set=7 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/16000 a=fmtp:101 0-15 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 SDP offer from CM: v=0 o=- 1543832047 2 IN IP4 10.112.23.41 s=- c=IN IP4 10.112.23.44 b=AS:64 t=0 0 m=audio 6064 RTP/AVP 8 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=ptime:20 CM sends a=rtpmap:101 telephone-event/8000 to a=rtpmap:101 telephone-event/16000 which is against RFC 4733: " The RTP payload format for named telephone events is designated as "telephone-event", the media type as "audio/telephone-event". In accordance with current practice, this payload format does not have a static payload type number, but uses an RTP payload type number established dynamically and out-of-band. The default clock frequency is 8000 Hz, but the clock frequency can be redefined when assigning the dynamic payload type. Named telephone events are carried as part of the audio stream and MUST use the same sequence number and timestamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway. The named telephone-event payload type can be considered to be a very highly-compressed audio codec and is treated the same as other codecs. "

If bitrate 16000 is not supported the correct response should be:

a=rtpmap:96 telephone-event/8000

Cause

Product defect

Solution

Fix will be available in CM 8.1

• About Avaya
• Contacts
• Careers
• Site Map

• Terms of Use
• Privacy Statement

• © 2026 Avaya LLC