[Spce-user] MS Lync with SipWise

[Chris Hoffmann]

Hi,

Over the last few weeks I have been experimenting with SipWise. I have
attempted to set up lync as a subscriber and can make calls with 2 way
audio however Lync does not appear to be reciving the SIP200 message as the
interface still shows ringing. I have done a number of wireshark captures
from the Lync server and compared calls via Asterisk which work and calls
via SipWise which have the above issue.


SIP 200 that doesn't get recognised by Lync
---------------------------------------------
SIP/2.0 200 OK
Record-Route: <sip:127.0.0.1:5062
;lr=on;ftag=6210295535;did=1c9.d452;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=anl6aVVKfUJ5RXlpRmYJGDtnIxIzMyx2HhVwEzN6DR4kF0UZBlsNMQ-->
Record-Route:
<sip:127.0.0.1;r2=on;lr=on;ftag=6210295535;ngcplb=yes;socket=tcp:
192.168.152.30:5060>
Record-Route:
<sip:192.168.152.30;transport=tcp;r2=on;lr=on;ftag=6210295535;ngcplb=yes;socket=tcp:
192.168.152.30:5060>
FROM: "Chris Hoffmann"<sip:+6490000431 at test.com
;user=phone>;epid=31D1C3F091;tag=6210295535
TO: <sip:+6421000004 at 192.168.152.30
;user=phone>;tag=3B920B2B-5AB35EC4000D82E5-2232B700
CSEQ: 52 INVITE
CALL-ID: 80348062-6b75-444f-a5b0-37ae28e958c1
VIA: SIP/2.0/TCP 192.168.152.18:50382;rport=50382;branch=z9hG4bK601318c1
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, OPTIONS, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 231
Contact: <sip:ngcp-lb at 192.168.152.30:5060
;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31>

v=0
o=dcom 1521704645 1521704648 IN IP4 192.168.152.30
s=SIP Call
c=IN IP4 192.168.152.30
t=0 0
m=audio 30300 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=direction:both
a=sendrecv
a=rtcp:30301

-------------------------------------------------

SIP 200 that works
-------------------------------------------------

SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.152.18:50391
;branch=z9hG4bK82fc4379;received=192.168.152.18
From: "Chris Hoffmann"<sip:+6490000431 at test.com
;user=phone>;epid=31D1C3F091;tag=9279e5c630
To: <sip:+64800000000 at 192.168.152.6;user=phone>;tag=as50ee3235
Call-ID: 6dd401d6-8bbc-4fcb-9f37-303d64293725
CSeq: 55 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:+64800000000 at 192.168.152.6;transport=TCP>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 830409115 830409116 IN IP4 192.168.152.6
s=Asterisk PBX 1.6.2.11
c=IN IP4 192.168.152.6
t=0 0
m=audio 10070 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---------------------------------------------------

Thanks,



Chris Hoffmann
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