[Asterisk-Users] Unknown RTP codec 72 received

[Elman Efendiyev]

Hi all,

I get "Unknown RTP codec 72 received" message in console when call in
progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN
over voicepulse connect (IAX) and to FWD echo test (SIP). But this
message only with one SIP client, others (X-Lite too) not giving this
message. All X-Lite settings are identical. Asterisk is last cvs version

This what I see in console (FWD):

vgw3*CLI>
    -- Executing Dial("SIP/332-552e", "SIP/613 at fwd") in new stack
    -- Called 613 at fwd
    -- SIP/fwd-357f is ringing
    -- SIP/fwd-357f answered SIP/332-552e
    -- Attempting native bridge of SIP/332-552e and SIP/fwd-357f Sep 13
11:02:52 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72
received Sep 13 11:02:57 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown
RTP codec 72 received Sep 13 11:03:01 NOTICE[245776]: rtp.c:489
ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:03:02
NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received
Sep 13 11:03:07 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP
codec 72 received
  == Spawn extension (full-access, 17009999613, 1) exited non-zero on
'SIP/332-552e'

And voicepulse:
vgw3*CLI>
    -- Executing Dial("SIP/332-3f30",
"IAX2/MPb32ouj46 at voicepulse/011XXXXXXXXXXXX") in new stack
    -- Called MPb32ouj46 at voicepulse/011XXXXXXXXXXXX
    -- Call accepted by 66.234.228.160 (format GSM)
    -- Format for call is GSM
    -- IAX2/voicepulse/3 stopped sounds
    -- IAX2/voicepulse/3 stopped sounds
    -- IAX2/voicepulse/3 answered SIP/332-3f30
Sep 13 11:06:37 NOTICE[262160]: rtp.c:489 ast_rtp_read: Unknown RTP
codec 72 received Sep 13 11:06:42 NOTICE[262160]: rtp.c:489
ast_rtp_read: Unknown RTP codec 72 received --------------------same
skipped--------------------------------------------------------------
Sep 13 11:10:24 NOTICE[262160]: rtp.c:489 ast_rtp_read: Unknown RTP
codec 72 received Sep 13 11:10:29 NOTICE[262160]: rtp.c:489
ast_rtp_read: Unknown RTP codec 72 received
    -- Hungup 'IAX2/voicepulse/3'
  == Spawn extension (full-access, 492103998463, 1) exited non-zero on
'SIP/332-3f30'

My sip.conf (234 user didn't give Unknown RTP codec 72 received message,
332 gives this message, only difference is internet path to users and
firewall type):

[general]
port = 5060
tos=lowdelay
videosupport=no
disallow = all
allow = gsm
allow = ulaw
canreinvite = no

[fwd]
context = in
type = peer
disallow = gsm
allow = ulaw
userneme = XXXXXX
secret = xxx
host = fwd.pulver.com

[234]
context = full-access
type = friend
disallow = ulaw
insecure = no
username = 234
secret = xxx
host = dynamic
nat = yes
dtmfmode = rfc2833
callerid = <234>

[332]
context = full-access
type = friend
disallow = ulaw
insecure = no
username = 332
secret = xxx
host = dynamic
nat = yes
dtmfmode = rfc2833
callerid = <332>

Could somebody tell me whay this "Unknown RTP codec 72 received" means
and how to fix it? Thanks.

--
Sincerely,
Elman Efendiyev
elman at earlinvest.com