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

Tue Sep 10 13:05:20 UTC 2019

It would appear you're calling yourself.

/b

On Tue, Sep 10, 2019 at 3:27 AM 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
>
>
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