[Spce-user] Issue with external to external transfer SIPWise 4.5.6

Robert Cuaresma rcuaresma at telcon.es
Thu Feb 1 13:00:02 EST 2018


Hi to all!

I have a problem with transfer with two external calls. I try to explain it:

The PBX is a Alcatel-Lucent OXO (last release) behind NAT with no issues 
with ingoing and outgoing calls. The only problem what I have is with 
external call transfer.

PBX OUTGOING CALL  =========> SIPWISE =========> PEER =========> 
EXTERNAL PHONE A  (GSM)
                                                        \
                                                 ATTENDED TRANSFER OK 
WITH FULL AUDIO.
                                                         /
PBX OUTGOING CALL  =========> SIPWISE =========> PEER =========> 
EXTERNAL PHONE B (GSM)   --> AFTER 60 SECONDS SEND BYE

(This trace was captured from call to Phone B)
Inital INVITE of call to Phone B:

     Request-Line: INVITE 
sip:62026XXXX at mydomaing.com;transport=UDP;user=phone SIP/2.0
     Message Header
         Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, UPDATE
         Supported: from-change,timer,histinfo
         User-Agent: OXO020/038.001 GW_020/084.001
         Session-Expires: 1800
         P-Asserted-Identity: <sip:93737XXXX at 10.0.92.246;user=phone>
         History-Info: 
<sip:62026XXXX at mydomaing.com;transport=UDP;user=phone>;index=1
         To: <sip:62026XXXX at mydomaing.com;user=phone>
         From: 
<sip:93737XXXX at 10.0.92.246;user=phone>;tag=df2bc741f74a56e6935582ead9ca0e77
         Contact: <sip:93737XXXX at 10.0.92.246;transport=UDP;user=phone>
         Content-Type: application/sdp
         Call-ID: 8cd732f6d8bedbd1cd896281fc5e46d3 at 10.0.92.246
         CSeq: 278499334 INVITE
         Via: SIP/2.0/UDP 
10.0.92.246;rport;branch=z9hG4bKf7234c35c21f59ee34c67a575e06cc84
         Max-Forwards: 70
         Content-Length: 268
     Message Body
         Session Description Protocol
             Session Description Protocol Version (v): 0
             Owner/Creator, Session Id (o): OxO 1517491417 1517491417 IN 
IP4 10.0.92.246
             Session Name (s): Alcatel-Lucent OXO020/084.001
             Connection Information (c): IN IP4 10.0.92.246
             Time Description, active time (t): 0 0
             Media Description, name and address (m): audio 32022 
RTP/AVP 18 106
             Media Attribute (a): sendrecv
             Media Attribute (a): rtpmap:18 G729/8000
             Media Attribute (a): fmtp:18 annexb=no
             Media Attribute (a): rtpmap:106 telephone-event/8000
             Media Attribute (a): fmtp:106 0-15
             Media Attribute (a): maxptime:60

And SIPWise Reply with 200 OK:

     Status-Line: SIP/2.0 200 OK
     Message Header
         Record-Route: 
<sip:127.0.0.1:5062;lr=on;ftag=df2bc741f74a56e6935582ead9ca0e77;did=c92.3e;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=ajRid25nc25iZXdhZnppTVlIamRqNGJ3blwGHQcXShEOFQco>
         Record-Route: 
<sip:127.0.0.1:5060;nat=yes;ngcplb=yes;r2=on;socket=udp:172.16.20.7:5060;ftag=df2bc741f74a56e6935582ead9ca0e77;lr=on>
         Record-Route: <sip:SIPWise Public 
IP:5060;nat=yes;ngcplb=yes;r2=on;socket=udp:172.16.20.7:5060;ftag=df2bc741f74a56e6935582ead9ca0e77;lr=on>
         To: 
<sip:62026XXXX at mydomaing.com;user=phone>;tag=0FD06CD1-5A73138500024518-37CFC700
         From: 
<sip:93737XXXX at 10.0.92.246;user=phone>;tag=df2bc741f74a56e6935582ead9ca0e77
         Call-ID: 8cd732f6d8bedbd1cd896281fc5e46d3 at 10.0.92.246
         CSeq: 278499334 INVITE
         Via: SIP/2.0/UDP 10.0.92.246;received=PBX PUBLIC 
IP;rport=56721;branch=z9hG4bKf7234c35c21f59ee34c67a575e06cc84
         Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, SUBSCRIBE, NOTIFY, 
PUBLISH, MESSAGE
         Supported: replaces
         Content-Type: application/sdp
         Content-Length: 290
         Contact: <sip:ngcp-lb at SIPWise Public 
IP:5080;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31>
     Message Body
         Session Description Protocol
             Session Description Protocol Version (v): 0
             Owner/Creator, Session Id (o): root 1435768594 1435768594 
IN IP4 SIPWise Public IP *<======= This is the IP of SIPWise*
             Session Name (s): Server
             Connection Information (c): IN IP4 SIPWise Public IP
             Time Description, active time (t): 0 0
             Media Description, name and address (m): audio 31586 
RTP/AVP 18 106
             Media Attribute (a): rtpmap:18 G729/8000
             Media Attribute (a): fmtp:18 annexb=no
             Media Attribute (a): rtpmap:106 telephone-event/8000
             Media Attribute (a): fmtp:106 0-16
             Media Attribute (a): maxptime:230
             Media Attribute (a): direction:both
             Media Attribute (a): sendrecv
             Media Attribute (a): rtcp:31587

When I press the Transfer key on the phone the PBX send to SipWise an 
INVITE without SDP:

Request-Line: INVITE sip:ngcp-lb at SIPWise Public 
IP:5080;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31 
SIP/2.0
         Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, UPDATE
         Supported: from-change,timer
         User-Agent: OXO020/038.001 GW_020/084.001
         Session-Expires: 1800;refresher=uac
         P-Asserted-Identity: <sip:9373XXXX at 10.0.92.246;user=phone>
         Contact: <sip:9373XXXX at 10.0.92.246;transport=UDP;user=phone>
         To: 
<sip:620266XXX at mydomain.com;user=phone>;tag=0FD06CD1-5A73138500024518-37CFC700
         From: 
<sip:93737XXXX at 10.0.92.246;user=phone>;tag=df2bc741f74a56e6935582ead9ca0e77
         Call-ID: 8cd732f6d8bedbd1cd896281fc5e46d3 at 10.0.92.246
         CSeq: 278499335 INVITE
         Via: SIP/2.0/UDP 
10.0.92.246;rport;branch=z9hG4bK8a538a45ad16cab680c0dde27fd3d7ac
         Max-Forwards: 70
         Content-Length: 0

And SIPWise reply with 200 OK with SDP. I see that, after this INVITE, 
_Sipwise has been changed the Media IP with the Media IP of the VoIP 
Service Provider_. This is wrong, because the Media IP should be the IP 
address of SipWise. _I have activated RTP Proxy with Plain SDP. _

Status-Line: SIP/2.0 200 OK
(...)
         Session Description Protocol
             Session Description Protocol Version (v): 0
             Owner/Creator, Session Id (o): root 1435768594 1435768595 
IN IP4 VoIP Service Provider IP <=============
             Session Name (s): Server
             Connection Information (c): IN IP4 VoIP Service Provider IP
             Time Description, active time (t): 0 0
             Media Description, name and address (m): audio 8542 RTP/AVP 
18 106
             Media Attribute (a): rtpmap:18 G729/8000
             Media Attribute (a): fmtp:18 annexb=no
             Media Attribute (a): rtpmap:106 telephone-event/8000
             Media Attribute (a): fmtp:106 0-16
             Media Attribute (a): sendrecv
             Media Attribute (a): maxptime:230
             Media Attribute (a): direction:both

Then, PBX send an ACK with SDP:

Request-Line: ACK sip:ngcp-lb at SIPWise Public 
IP:5080;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31 
SIP/2.0
Message Body
     Session Description Protocol
         Session Description Protocol Version (v): 0
         Owner/Creator, Session Id (o): OxO 1517491417 1517491418 IN IP4 
10.0.92.246
         Session Name (s): Alcatel-Lucent OXO020/084.001
         Connection Information (c): IN IP4 10.0.92.246
         Time Description, active time (t): 0 0
         Media Description, name and address (m): audio 32028 RTP/AVP 18 106
         Media Attribute (a): fmtp:18 annexb=no
         Media Attribute (a): rtpmap:18 G729/8000
         Media Attribute (a): rtpmap:106 telephone-event/8000
         Media Attribute (a): fmtp:106 0-15
         Media Attribute (a): sendrecv
         Media Attribute (a): ptime:30
         Media Attribute (a): maxptime:120

After this, the RTP flow is send directly from PBX to VoIP Service 
Provider instead of send RTP flow to SIPWise, how it should be. 60 
seconds (exactly) later, the PBX send a BYE and hangup the call with 
Phone A.

In conclusion, on that scenario i have one leg of the call (Phone A) 
sendly RTP Flow to SipWise and another leg of the call (Phone B) sendly 
RTP flow directly to the VoIP Service Provider...

I have made some tests and if I configure Subscriber with RTP Proxy 
defined to "Never", the call no hangup after 60 seconds and works fine, 
but the RTP Flow goes always from customer PBX  directly to VoIP Service 
Provider. What would be the cause of this issue?

I apreciate so much your help!

-- 
Saludos,
Robert Cuaresma
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