[Asterisk-Users] RTP codec 13 received - Cisco incompatibilit y?

[Skuse, Phil]

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:cerrajetto at pyme.net]
Sent: 31 July 2003 09:09
To: asterisk-users at lists.digium.com
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 300 at 192.168.200.200 (Extension 300 in 
Asterisk)
 - Asterisk transfers to 666554433 at 192.168.200.99 (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/666554433 at 192.168.200.200")

in new stack
    -- Called 666554433 at 192.168.200.200
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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