[OpenSIPS-Users] Calls being disconnected

SamyGo govoiper at gmail.com
Tue Jul 3 13:18:18 CEST 2012


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
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> _______________________________________________
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>>> Users at lists.opensips.org
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>>>
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