[Freeswitch-users] Two calls with same call-id

Brian West brian at freeswitch.com
Thu Sep 12 17:26:25 UTC 2019


Show me a pcap?

On Tue, Sep 10, 2019 at 9:06 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> What do you mean exactly by “managed”?
>
> They are not the same call. Every time a call is created in FS, FS created
> a unique UUID, and that’s what is used by Fe to manage the call I question.
>
> So when you create an outbound call. It has an associated UUID, and when
> the same call is sent to you by the proxy, it created another UUID.
>
> You treat them as different calls altogether.
>
> On Tue, 10 Sep 2019 at 10:35, Dominique Jeannerod <
> dominique.jeannerod at interactiv-group.com> wrote:
>
>>   Hi all,
>>
>> currently working to understand some calls without RTP (one-way audio, or
>> no audio at all), I noticed a strange case in our traces : we send an
>> outgoing call on a trunk. The destination number is translated to another
>> number, and the call comes back to our SBC (the new dest is on our network).
>> What i noticed is that the incoming call has the same call-id as our
>> outgoing call (I suppose that the other operator uses proxies rather than
>> SBCs), and it comes to the *same *freeswitch (has 2 IP addresses)
>>
>> So, wondering what could be the impact of that on our Freeswitch SBC.
>> Can those 2 calls be managed as one and only call by freeswitch ?
>>
>> Thanks with anticipation
>> Best regards,
>>
>> Dominique Jeannerod
>>
>> *Calls traces : *
>> *First call (outgoing) :* Our SBC IP1 -> External IP 1
>> INVITE sip:+<Dest number 1>@<External IP 1> SIP/2.0
>> Via: SIP/2.0/UDP <Our SBC IP 1>;rport;branch=z9hG4bKg6y1BXQ5ZvreF
>> Max-Forwards: 68
>> From: "Source num" <sip:+<Source Num>@< Our SBC IP 1>;tag=r8p71798metHN
>> To: <sip:+ <Dest number 1>@<External IP 1>
>>
>> *Call-ID: d3b11cd5-4a87-1238-068f-5254000db631*CSeq: 9292108 INVITE
>> Contact: <sip:gw+gwname@ <Our SBC IP 1>:5060;transport=udp;gw=gwname>
>> User-Agent: FreeSWITCH
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
>> NOTIFY
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, refer
>> Session-Expires: 14400;refresher=uac
>> Min-SE: 120
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 248
>> X-FS-Support: update_display,send_info
>> P-Asserted-Identity: "<PAI Display>" <sip:+<PAI Number>@ <Our SBC IP 1> >
>> v=0
>> o=FreeSWITCH 1567668560 1567668561 IN IP4 <Our SBC IP 1>
>> s=FreeSWITCH
>> c=IN IP4 <Our SBC IP 1>
>> t=0 0
>> m=audio 23240 RTP/AVP 8 101 13
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP <Our SBC IP 1> ;received= <Our SBC IP 1>
>> ;rport=5060;branch=z9hG4bKg6y1BXQ5ZvreF
>> From: "Source num" <sip:+<Source Num>@< Our SBC IP 1>>;tag=r8p71798metHN
>> To: <sip:+3 Dest number 1>@<External IP 1>;tag=t25KQD4evZ2DF
>> Call-ID: d3b11cd5-4a87-1238-068f-5254000db631
>> CSeq: 9292108 INVITE
>> Contact: <sip:+<Dest number 1>@ <External IP 1>;did=4a5.72bd7e82>
>> User-Agent: FreeSWITCH
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
>> NOTIFY
>> Require: timer
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, refer
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 247
>> X-FS-Display-Name: Outbound Call
>> X-FS-Display-Number: sip: <Dest number 1>@<domain>
>> X-FS-Support: update_display,send_info
>> v=0
>> o=FreeSWITCH 1567660096 1567660097 IN IP4 <Our SBC IP 2>
>> s=FreeSWITCH
>> c=IN IP4 <External RTP IP 1>
>> t=0 0
>> m=audio 58152 RTP/AVP 8 101 13
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>>
>> *Second call (Incoming) :* External IP2 -> Our SBC IP 2
>> INVITE sip:+<Dest number 2>@<Our SBC IP 2>:5060;user=phone SIP/2.0
>> Via: SIP/2.0/UDP <External IP 2>:5060;branch=z9hG4bK446.1e47ee86.0
>> Max-Forwards: 62
>> From: "Source num" <sip:+<Source
>> Num>@<domain>;user=phone>;tag=r8p71798metHN
>> To: <sip:+<Dest number 2>@<domain>;user=phone>
>>
>> *Call-ID: d3b11cd5-4a87-1238-068f-5254000db631*CSeq: 9292108 INVITE
>> Contact: <sip:gw+<gwname>@<External IP 2>;did=4a5.e4c51b72>
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
>> NOTIFY
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, refer
>> Session-Expires: 14400;refresher=uac
>> Min-SE: 120
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 247
>> X-FS-Support: update_display,send_info
>> P-Asserted-Identity: "<PAI Display>" <sip:+<PAI
>> Number>@<domain>;user=phone>
>> v=0
>> o=FreeSWITCH 1567668560 1567668561 IN IP4 <Our SBC IP 1>
>> s=FreeSWITCH
>> c=IN IP4 <External RTP IP 2>
>> t=0 0
>> m=audio 56204 RTP/AVP 8 101 13
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP <External IP 2>:5060;branch=z9hG4bK446.1e47ee86.0
>> From: "Source num" <sip:+<Source
>> Num>@<domain>;user=phone>;tag=r8p71798metHN
>> To: <sip:+<Dest number 2>@<domain>;user=phone>;tag=t25KQD4evZ2DF
>> Call-ID: d3b11cd5-4a87-1238-068f-5254000db631
>> CSeq: 9292108 INVITE
>> Contact: <sip:+<Dest num 2>@>Our SBC IP 2>:5060;transport=udp>
>> User-Agent: FreeSWITCH
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
>> NOTIFY
>> Require: timer
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, refer
>> Session-Expires: 14400;refresher=uac
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 248
>> X-FS-Display-Name: Outbound Call
>> X-FS-Display-Number: sip:<Dest number 2>@keyyo.net
>> X-FS-Support: update_display,send_info
>> P-Asserted-Identity: "Outbound Call" <sip:<Dest number 2>@<domain>>
>> v=0
>> o=FreeSWITCH 1567660096 1567660097 IN IP4 <Our SBC IP 2>
>> s=FreeSWITCH
>> c=IN IP4 <Our SBC IP 2>
>> t=0 0
>> m=audio 31704 RTP/AVP 8 101 13
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>>
>>
>> Ce message et toutes les pièces jointes (ci-après le “message”) sont
>> établis à l’intention exclusive de ses destinataires et sont confidentiels.
>> Si vous recevez ce message par erreur, merci de le détruire et d’en avertir
>> immédiatement l’expéditeur. Toute utilisation de ce message non conforme a
>> sa destination, toute diffusion ou toute publication, totale ou partielle,
>> est interdite, sauf autorisation expresse. L’internet ne permettant pas
>> d’assurer l’intégrité de ce message . Interactiv-group (et ses filiales)
>> décline(nt) toute responsabilité au titre de ce message, dans l’hypothèse
>> ou il aurait été modifié. IT, ES, UK.
>> <https://interactiv-group.com/disclaimer.html>
>> _________________________________________________________________________
>>
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>>
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>>
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>>
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>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

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