[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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