asterisk users - Sep 2004 - Unknown RTP codec 72 received

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@fwd") in new
stack
    -- Called 613@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@voicepulse/011XXXXXXXXXXXX") in new stack
    -- Called MPb32ouj46@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@earlinvest.com

On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote:
> 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
None of the 19 hits I saw on Google about this were helpful?

<macro>To search the Asterisk mailing list archive go to www.google.com
and put site:lists.digium.com in addition to your other query
terms.</macro>

-- 
          Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

According to IANA's list of RTP payload types
(http://www.iana.org/assignments/rtp-parameters) RTP payload type 72 fulls
within the following range:

      72--76     reserved for RTCP conflict avoidance             [RFC3550]

I can't find much else in RFC3550 that defines it further but this should
start you on the right path I hope.

-----Original Message-----
From: Eric Wieling [mailto:eric@fnords.org]
Sent: 13 September 2004 16:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unknown RTP codec 72 received


On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote:
> 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


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19 Question + this one and no answer;

Does anyone have a clue what causes "Unknown RTP codec 72 received"
notice
and how to fix it?

Regards
Danny

> -----Original Message-----
> From: Danny Froberg [mailto:danny@froberg.org] 
> Sent: Sunday, October 24, 2004 4:37 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Unknown RTP codec 72 received
> 
> 
> 19 Question + this one and no answer;
> 
> Does anyone have a clue what causes "Unknown RTP codec 72 
> received" notice 
> and how to fix it?
> 
> Regards
> Danny
> 
I receive this message when I call from X-Lite.  I notice that
it is usually when I am sending DTMF digits.  I could be using 
the wrong dtmfmode (using info), but I am not sure.

This message is rather annoying so I would definitely like to 
see if anyone else has gotten it figured out.

Robert Jackson