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. 

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