[Asterisk-Users] RTP codec 13 received - Cisco incompatibility?

[Mark Spencer]

Probably needs some more information.  I would consider placing a detailed
bug report in the bug tracker including the output of "sip debug" with a
call going through.

Mark

On Thu, 31 Jul 2003, Cerrajetto wrote:

> 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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