[OpenSIPS-Users] Calls being disconnected

Wed Jul 4 19:15:15 CEST 2012

I’m trying to debug those calls using wireshark and VoIP calls analisys ... 

And i guess it might be a network problem, maybe some packets being dropped .. Is there any chance of being that? 
Because my Opensips box and my FreePBX (asterisk) they are in the same network, just passing through a switch, among with other servers (including another opensips who is giving me the same problem)

From: Rodrigo Ferreira 
Sent: Wednesday, July 04, 2012 9:10 AM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Calls being disconnected

That was my bad, its different IPs, but I wrote it wrong.

I will get some new traces today, and I will write the IPs correctly 

From: SamyGo 
Sent: Wednesday, July 04, 2012 1:54 AM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Calls being disconnected

I will definitely try to read the SIP traces. I love trying to solve the puzzle but the very first line !! 

U 201.71.XXX.XXX:5060 -> 201.71.XXX.XXX:5060

Is it the same server originating SIP INVITE sending to the same server !! if not please differentiate the two addresses like.

U 201.71.XXX.XXX:5060 -> 201.71.XXX.YYY:5060 

Goodness , its FBX,OpenSIPS, gatewaySTS, and even Asterisk 1.8 -- are they all using the same IP !!

Caller(FPBX)<--->OpenSIPS+RTPproxy<---->gatewaySTS<---->Asterisk
   ||<============================RTP===============>||

Please carefully use different IP addresses of all the SIP servers/UACs in your particular call scenario. What I'm still suspecting with more intensity is that somehow the RTPproxy is called but Asterisk and FPBX negotiate media directly in SDPs and bam..RTPproxy gets angry at this insulting attitude and disconnects the call !! That's my guess.

OpenSIPS logs along with the new SIP traces will help clarify the real cause.

Thanks
Sammy

On Tue, Jul 3, 2012 at 7:23 PM, Rodrigo Ferreira <rodrigo.ferreira at vipway.net.br> wrote:

  I’m attaching the log that I got from a call that was disconnected

  From: Rodrigo Ferreira 
  Sent: Tuesday, July 03, 2012 11:09 AM
  To: OpenSIPS users mailling list 
  Subject: Re: [OpenSIPS-Users] Calls being disconnected

  I’m getting the logs, but let me ask you something, force_rport will help me? 

  From: SamyGo 
  Sent: Tuesday, July 03, 2012 8:18 AM
  To: OpenSIPS users mailling list 
  Subject: Re: [OpenSIPS-Users] Calls being disconnected

  Specifically talking about one where call gets dropped - what is the network scenario !?  Make one such call and take a sip trace on the opensips server using tcpdump or sipgrep. Share the pcap. As well as the opensips logs please. 

  On Tue, Jul 3, 2012 at 4:12 PM, Rodrigo Ferreira <rodrigo.ferreira at vipway.net.br> wrote:

    I’ve a lot of different scenarios through my network, I have clients behind NAT, from other networks, because I work for a telephony company.

    From: SamyGo 
    Sent: Tuesday, July 03, 2012 8:02 AM
    To: OpenSIPS users mailling list 
    Subject: Re: [OpenSIPS-Users] Calls being disconnected

    RTPproxy - thats what I suspected. What i'm guessing is that RTPproxy is engaged but somehow the end points receive each other's Direct IPs in SDP...hmm...tell me if your both end points are on the same LAN and server is placed somewhere in internet !!

    On Tue, Jul 3, 2012 at 3:58 PM, Rodrigo Ferreira <rodrigo.ferreira at vipway.net.br> wrote:

      I will take a look at my rtp .. I’m using rtpproxy on my box, I will try to get some logs today too

      From: SamyGo 
      Sent: Tuesday, July 03, 2012 4:34 AM
      To: aamir chougule ; OpenSIPS users mailling list 
      Subject: Re: [OpenSIPS-Users] Calls being disconnected

      Besides logs and any sip traces as Aamir stated it'd be hard to pin point anything. One of the big reason for automatically dropping an established call specially after a specific amount of time i.e 30 seconds..thats because of lack of RTPs flowing through the server and if media-relaying tools are used they will send hangup to both ends, even though both ends have their RTP flowing directly.

      Regards, 
      Sammy

      On Tue, Jul 3, 2012 at 12:21 PM, aamir chougule <aamir_ryu at yahoo.com> wrote:

        Hi Rodrigo,

        The only which I know is after the call gets connected i.e. 200OK from the other end there should always be an ACK for that 200OK, if there is no ACK for the 200OK then the call will get disconnected after sometime.

        Can you please give us the logs for the call by tracing it through ngrep tool because without any sip traces or opensips log we can't tell what is going on within the box.

        Regards,

        Aamir Chougule
        Cell: 09167989111

------------------------------------------------------------------------
        From: Rodrigo Ferreira <rodrigo.ferreira at vipway.net.br>
        To: OpenSIPS users mailling list <users at lists.opensips.org> 
        Sent: Monday, 2 July 2012 10:40 PM
        Subject: [OpenSIPS-Users] Calls being disconnected

        Hey guys,

        I’m having problems with calls being “disconnected” and I dont know where I can start to look at, because on my CDR all those calls ends with a 200OK, but that isnt true, because you are talking all the suddenly the call stop.

        Any ideas where I should start looking at?

        Engº Rodrigo Ferreira
        Supervisor de Telefonia
        VIPWay Telecom
        Tel.: +55 13 4010-1000
        Cel.: +55 13 8136-5839

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