[OpenSIPS-Users] Calls being disconnected

Wed Jul 4 06:54:50 CEST 2012

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 <rodrigo.ferreira at vipway.net.br>
> *Sent:* Tuesday, July 03, 2012 11:09 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>
> *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 <govoiper at gmail.com>
> *Sent:* Tuesday, July 03, 2012 8:18 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>
> *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 <govoiper at gmail.com>
>> *Sent:* Tuesday, July 03, 2012 8:02 AM
>>  *To:* OpenSIPS users mailling list <users at lists.opensips.org>
>>  *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 <govoiper at gmail.com>
>>> *Sent:* Tuesday, July 03, 2012 4:34 AM
>>> *To:* aamir chougule <aamir_ryu at yahoo.com> ; OpenSIPS users mailling
>>> list <users at lists.opensips.org>
>>> *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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>>>>
>>>>
>>>>
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>>>>
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