[c-nsp] SIP to ISDN Call Progress

Brian Turnbow b.turnbow at twt.it
Mon Nov 15 07:51:49 EST 2010


Hi Marco

See inline

> -----Original Message-----
> From: cisco-nsp-bounces at puck.nether.net [mailto:cisco-nsp-
> bounces at puck.nether.net] On Behalf Of Marco Marzetti
> Sent: lunedì 15 novembre 2010 10:36
> To: cisco-nsp at puck.nether.net
> Subject: [c-nsp] SIP to ISDN Call Progress
> 
> 
> Hello,
> 
> I have a problem with SIP to ISDN internetworking on Cisco IOS.
> I'm unable to receive early-media messages from the ISDN side of the
> call.
> Hardware and software versions are: Cisco 2800 Software
> (C2800NM-ENTSERVICES-M), Version 12.4(20)T6, RELEASE SOFTWARE (fc2).
> 
> # debug isdn q931 int Se0/1/0:15
> Nov 15 10:06:54.437 CET: ISDN Se0/1/0:15 Q931: Applying typeplan for
> sw-type 0x12 is 0x0 0x1, Calling num 03631970XXX
> Nov 15 10:06:54.441 CET: ISDN Se0/1/0:15 Q931: Sending SETUP  callref =
> 0x0D0D callID = 0x980D switch = primary-net5 interface = User
> Nov 15 10:06:54.441 CET: ISDN Se0/1/0:15 Q931: TX -> SETUP pd = 8
> callref
> = 0x0D0D
> 	Bearer Capability i = 0x8090A3
> 		Standard = CCITT
> 		Transfer Capability = Speech
> 		Transfer Mode = Circuit
> 		Transfer Rate = 64 kbit/s
> 	Channel ID i = 0xA9839F
> 		Exclusive, Channel 31
> 	Calling Party Number i = 0x0180, '03631970XXX'
> 		Plan:ISDN, Type:Unknown
> 	Called Party Number i = 0x81, '199151119'
> 		Plan:ISDN, Type:UnknownsipSPIUpdateRtcpSession: sx79861:
> started RTP
> timer in state STATE_SENT_ALERTING
> 
> Nov 15 10:06:54.457 CET: ISDN Se0/1/0:15 Q931: RX <- SETUP_ACK pd = 8
> callref = 0x8D0D
> 	Channel ID i = 0xA9839F
> 		Exclusive, Channel 31
> 
> 
> Nov 15 10:06:56.745 CET: ISDN Se0/1/0:15 Q931: RX <- CALL_PROC pd = 8
> callref = 0x8D0D


There is no progress indicator indicating that inband information is now available.
So no opening of the audio channel. You can play around with the progress indicators you send out with your setup to manipulate this 
Try searching for no ringback , no busytones  on the cisco website for an in depth explanation.





> The router places the call to our public switch and cut-through the
> voice
> path only after the CONNECT message ignoring the CALL_PROC messages and
> the
> relative early-audio
> stream.
> Looking at the SIP side of the call no "SIP 183 Progress" is sent by
> the
> router between the "Trying" and the "Ringing" messages.
> 
> I would expect the router to generate proper SIP signaling and cut-
> through
> in the backward direction the voice path after the CALL_PROC message
> has
> been received.

Well there is no message saying that inband info is now available and  the router needs to know when to do it....
If you throw nat in the picture it can be fun getting the audio to the device once RTP is active as well, but that is another story!!

Brian


> 
> Any help?
> 
> Thank you
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