Skuse, Phil
2003-Jul-31 01:46 UTC
[Asterisk-Users] RTP codec 13 received - Cisco incompatibilit y?
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@pyme.net]
Sent: 31 July 2003 09:09
To: asterisk-users@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@192.168.200.200 (Extension 300 in
Asterisk)
- Asterisk transfers to 666554433@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@192.168.200.200")
in new stack
-- Called 666554433@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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Iain Stevenson
2003-Jul-31 03:49 UTC
[Asterisk-Users] RTP codec 13 received - Cisco incompatibilit y?
.. 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@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@pyme.net] > Sent: 31 July 2003 09:09 > To: asterisk-users@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@192.168.200.200 (Extension 300 in > Asterisk) > - Asterisk transfers to 666554433@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@192.168.200.200") > > in new stack > -- Called 666554433@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. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >