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TOPIC: SIP / VOIP via Cisco AS5300

SIP / VOIP via Cisco AS5300 11 years 3 months ago #8956

  • Ranger24
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Hi Guys,

I'm hoping you can help with a little VoIP problem I have. One of my customers has the following network:


POTS --> SIP DSLAM --> SWITCH --> AS5300 -->PSTN

The AS5300 has a connection to an IPTel Sip Server

(POTS = Plain Old Telephone System - analogue phone!)

When they make a SIP call: POTS -DSLAM - Switch - AS5300 - Switch DSLAM - POTS everything works well and as it should be.

When they make a call through to the PSTN they get some odd characteristics:

1) ringback tone (as heard by the calling party) lasts 9 seconds then stops

2) 20 seconds later the call is terminated.

On to my questions:

A) Can anyone suggest a fault methodology for me to isolate the fault?

B) Any tips on how to decipher a SIP debug on the AS5300 - I have 6 pages to go through!

c) Anyone have an experiance they can offer me.

My responsibility is the SIP DSLAM & I need really to isolate the fault as either the DSLAM or the CISCO, then provide [suggest] a fix.

Thanks for any, and all help.

br

Ranger

Patience - the last reserve of the any engineer
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Re: SIP / VOIP via Cisco AS5300 11 years 3 months ago #8976

  • Chris
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Tough problem Ranger.

I'll have to look into it and see what I can come up with. I'm not sure what the chances are finding a solution for it, but I'll give it a go in hope for something that can help you resolve it.
Chris Partsenidis.
Founder & Editor-in-Chief
www.Firewall.cx
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further info 11 years 3 months ago #8986

  • Ranger24
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After a little effort and a highlighter I think the problem is found, if not understood.

Basically the server is interrupting the session setup phase, and failing to resolve the SIP phones messages correctly:

1) Sip client A send INVITE to server.

2) Server returns INVITE to client A with WWW-Authenticate field

3) Client A send ACK message in response to 2)

4) Client A retransmits Invite, with Authentication in message body

5) Server forward invite to PSTN GW for client B (PSTN phone)

6) 100 trying... message sent: GW --> Server --> Client A

(All is correct so far)

7) GW sends "183 Session Progress" message to server, Server forwards to client A. Contains "100rel" field indicating reliable transport to be used (TCP) for the reply

( As I understand it 183 indicates session traffic is ready to be sent to client A before session established. I believe this indicates that the ringing signal from the PSTN has arrived at GW before session set-up is complete. I'd expect this to happen)

8) CLient A responds with PRACK message which server forwards to GW.

9) GW responds with "500 INTERNAL SERVER ERROR" message

10) GW then closes the session with a "200 OK" message to client A

I believe there are 2 faults here:

1) The handling of the PSTN ringing by the GW. It should recieve this anc convert it to "180 ringing" message NOT "183 Session Progress"

2) The server does not appear to know how to handle PRACK messages (RFC 3262). Hence the "500 Internal Server error"

br

Ranger

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