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

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