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

[Iain Stevenson]

.. 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" 
<Phil.Skuse at vicorp.com> 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: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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