.. poking head above parapet, venturing correction ..
RTP payload type 13 is "comfort noise" viz
<http://www.iana.org/assignments/rtp-parameters>
whereas payload type 19 is "reserved". Maybe Cisco is right ;-)
I believe * has a partial implementation of comfort noise but that it's not complete yet. I found I could ignore the error messages with my Cisco ATA 186s.
Iain
--On Thursday, July 31, 2003 9:46 am +0100 "Skuse, Phil" <[EMAIL PROTECTED]> wrote:
I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about this.
My system does not have the problem you describe. I can call from a SIP softphone, through asterisk , through the cisco and out to our meridian system or the PSTN. In fact, it works very well. Are you sure that you have the dial-peers on the router configured correctly?
-----Original Message----- From: Cerrajetto [mailto:[EMAIL PROTECTED] Sent: 31 July 2003 09:09 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RTP codec 13 received - Cisco incompatibility?
Hello,
In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9.
If a client softphone calls directly via Cisco to the PSTN, the call works successfully.
If the client softphone calls via Asterisk to other SIP internal extension, it work fine too.
The problem is when a client calls an Asterisk extension, and Asterisk transfers the call (via SIP) to the Cisco:
- Pingtel (192.168.1.10) calls [EMAIL PROTECTED] (Extension 300 in Asterisk) - Asterisk transfers to [EMAIL PROTECTED] (Cisco GW) - Cisco tries to call to PSTN (666554433)
In that context, Asterisk generates this message while ringing:
NOTICE[540685]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 received
The PSTN recipient's phone rings. The client does not receive the typical intermittent tone/signal that means "the recipient's phone is ringing". When
the recipient answers, the call is inmediantly finished. Maybe a short "Hello" can be listened.
Asterisk shows a response back from Cisco:
Bad Request - 'Invalid IP Address'
In sip.conf, Asterisk is forced to use g711ulaw. I've tried other codecs with no success.
What is the real problem?. Is it a RTP problem with "codec 13", o a SIP problem?. Is there a Cisco-Asterisk incompatibility?.
This is the sequence generated by Asterisk:
-- Registered SIP 'pingtel01' at 192.168.1.10 port 5061 expires 500 -- Executing Dial("SIP/pingtel01-af0d", "SIP/[EMAIL PROTECTED]")
in new stack -- Called [EMAIL PROTECTED] NOTICE[573453]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 received NOTICE[573453]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 received -- SIP/192.168.200.200-a3d2 answered SIP/pingtel01-af0d -- Attempting native bridge of SIP/pingtel01-af0d and SIP/192.168.200.200- a3d2 -- Got SIP response 400 "Bad Request - 'Invalid IP Address'" back from 192.168.200.99 == Spawn extension (default, 003, 1) exited non-zero on 'SIP/peter-af0d'
Thank you very much, Mark Cerrajetto.
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