ast_rtp_read: Unknown RTP codec X received

11.5.1.1. ast_rtp_read: Unknown RTP codec X received

1) Unknown RTP codec 96 received:

If you are getting messages such as the one below, chances are big you are trying to connect to a Cirpack/v4.39a (gw_sip) sip gateway or to a Mediatrix 1124 gateway

Dec 24 10:57:40 NOTICE[5570]: rtp.c:564 ast_rtp_read: Unknown RTP codec 96 received

This is because those 2 devices for some reason decide to send DTMF as a dynamic payload type 96, and asterisk expects the DTMF to have number 101 or 121.

We can easily fix this with a one line patch to rtp.c

Look for:

[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[97] = {1, AST_FORMAT_ILBC},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},

and change this to:

[34] = {1, AST_FORMAT_H263},
[103] = {1, AST_FORMAT_H263_PLUS},
[96] = {0, AST_RTP_DTMF},
[97] = {1, AST_FORMAT_ILBC},
[101] = {0, AST_RTP_DTMF},
[110] = {1, AST_FORMAT_SPEEX},

now recompile asterisk, and everything should work fine.

References:

http://www.asteriskguru.com/archives/asterisk-users-asterisk-devel-mediatrix-dtmf-bug-solved-vt13784.html
http://www.ethereal.com/lists/ethereal-users/200408/msg00043.html
http://www.freesoft.org/CIE/RFC/1890/29.htm
http://www.iana.org/assignments/rtp-parameters

2) Unknown RTP codec 57 received:

seems to happen on SPA
http://www.asterisknerds.com/logs/irclogger_log/asterisk?date=2005-05-05,Thu&text=on

3) Unknown RTP codec 72 received:

I've never seen this one myself, but by googling around this is probably:

1) you are using info for DTMF with xlite. (unconfirmed)

2) you are using an asterisk version older than 12-06-04 and have allow=all (which means you should upgrade)

3) Most probable: something is sending RTCP packets in the RTP datastream to asterisk and asterisk doesn't know what to do with it.

To get rid of this, turn of symmetric RTP. (turn NAT to no, but still use externip if you are using the sip client behind a nat).
If you are using xlite, you can also try to turn it off in the phone settings,

the setting is in Main Menu > Advanced System Settings > RTP Settings > Obey Reverse UDP Mapping Rules = NO.
and
Xten menu > Advanced Settings > RTP Settings > Send RTCP messages = NO

You might need to use stun to make it work without symmetric RTP.

references:

http://bugs.digium.com/bug_view_advanced_page.php?bug_id=2945
http://www.asteriskguru.com/archives/asterisk-users-unknown-rtp-codec-72-received-vt21186.html
http://www.asteriskguru.com/archives/asterisk-users-error-unknown-rtp-codec-72-received-vt27771.html
http://www.ietf.org/rfc/rfc3550.txt
http://voxilla.com/forum-viewtopic-t-4475.html

4) Unknown RTP codec 19 received:

This one seems to be a problem similar to unknown codec 96 received (see above).
It appears when DTMF is being sent on the PSTN phone connected to a cisco as5350, and sent to asterisk over RTP Named Telephone Event (RFC 2833).

The solution is probably a patch as in the unknown codec 96, or reconfiguring the cisco gateway.

(please let us know how you fixed this, if you encounter this problem)

references:

https://puck.nether.net/pipermail/cisco-voip/2004-February/000375.html

5) Unknown RTP codec 85 received:

I only found one reference to this problem, possibly a problem where allow=all is used. (don't use that)
If that doesn't fix it, try upgrading your asterisk, chances are big the problem is already fixed.

If you experience this problem and can or cannot fix it, please let us know!!!!

references:

http://www.marko.net/asterisk/archives/0302/0035.html

6) Unknown RTP codec 100 received:

You probably connected some fax to some ATA, and then try to send faxes over t.30 to asterisk, this is not good, this fails in almost all cases.

The real solution is to use a t.38 enabled ATA and patch asterisk to use t.38.

references:

http://bugs.digium.com/view.php?id=5090

7) Lots of lines with different unknown codecs received:

Same as the problem above, you probably tried to connect some fax to an ATA, this is what might happen during the fax training.

ATA's are not made to be connected to faxes unless they are t.38 capable.

If it doesn't work, give up now, its not meant to be used like that.

------------------

If you know of any other unknown codec errors, please post them here. (and if you find the solution later, come back and post the solution please!!!!).

User Comments

Svarog (uros dot djokic at gmail dot com)
04 May 2008 14:39:36
I have the same problem as tetsuo191 with codec 126.

tetsuo191 (tetsuo2k6 at web dot de)
28 March 2008 10:02:26
I get this from time to time (from different IP's in our LAN):

[Mar 28 09:52:25] NOTICE[5328]: rtp.c:1287 ast_rtp_read: Unknown RTP codec 126 received from '192.168.0.14'
[Mar 28 09:52:32] NOTICE[5352]: rtp.c:1287 ast_rtp_read: Unknown RTP codec 126 received from '192.168.0.33'
[Mar 28 09:52:35] NOTICE[5328]: rtp.c:1287 ast_rtp_read: Unknown RTP codec 126 received from '192.168.0.14'

Doesn't harm the telephony though, I just wonder what it is.... codec 126 doesn't seem to be defined in the RTP standart (at least that's what wikipedia says)

IdioT Staffing Services (IdioTStaffingServices at gmail dot com)
12 November 2006 05:22:04
post deleted because of SPAM. Stupid bastards.

Roberto Braga (roberto at talktelecom dot com dot br)
09 November 2006 19:22:08
Using dtmfmode=inband whith codec G729A.
Edit dsp.c and search for the first line where the word "codec" appears. Is de "default:" section of a switch statement, comment it out and copy and paste the ALAW section right above.
Recompile Asterisk and install it. Now you can send DTMF digits INBAND using codec G729A.
I hope this helps.

Ken (kwolf at iafrate dot com)
30 October 2006 17:31:13
We are experiencing this error but can make and take calls, just no audio at all. Any Ideas??
Oct 30 10:59:27 NOTICE[5258]: rtp.c:564 ast_rtp_read: Unknown RTP codec 2 received

astnic (astnic20 at yahoo dot fr)
10 October 2006 15:02:33
hi,
i'm receiving
unknown rtp codec 126 received.
i don't know how to fix it.

rrgbr (news at rrgbr dot de)
26 April 2006 01:25:46
change rtp.c and compile asterisk to activate the codec:

//#ifdef USE_DEPRECATED_G726
[2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
//#endif

xeron (shrivastaw at gmail dot com)
05 January 2006 09:51:23
Hi I am receving
Unknown RTP codec 2 received
when tryint to use g.726 in my sip.conf

allow=g726

and call is being made, though my end users are connected still they cant talk and this error fills up the asterisk console.

Please help

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