[Spce-user] How to force rtpengine for peer?

[Serge S. Yuriev]

Hello,

Our SPCE (x.x.x.142) installed in DMZ so all of our local customers 
(10.0.64) get there with theirs original rfc1918 addresses w/o any NAT.

After routing SPCE sends
U 2015/04/08 16:47:13.700009 x.x.x.142:5060 -> y.y.y.28:5060
INVITE sip:7757#74996383868 at y.y.y.28:5060;transport=udp SIP/2.0'
Record-Route: 
<sip:x.x.x.142;r2=on;lr=on;ftag=44CA22C8-55253161000AACCF-D6205700;ngcplb=yes>'
Record-Route: 
<sip:127.0.0.1;r2=on;lr=on;ftag=44CA22C8-55253161000AACCF-D6205700;ngcplb=yes>'
Via: SIP/2.0/UDP 
x.x.x.142;branch=z9hG4bKc901.e9c1f6b9722f8668caa9aba4a1dbf968.0'
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKNhXMQaha;rport=5080'
From: <sip:12550 at 10.0.64.64>;tag=44CA22C8-55253161000AACCF-D6205700'
To: <sip:7757#74996383868 at y.y.y.28>'
CSeq: 10 INVITE'
Call-ID: c168cc00-52513161-2fb71-4040000a at 10.0.64.64'
Max-Forwards: 69'
Supported: resource-priority,replaces'
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, 
SUBSCRIBE, NOTIFY'
Expires: 180'
Supported: X-cisco-srtp-fallback,X-cisco-original-called'
Call-Info: 
<sip:10.0.64.64:5060>;method="NOTIFY;Event=telephone-event;Duration=500"'
P-Asserted-Identity: <sip:12550 at 10.0.64.64>'
Content-Length: 0'
Contact: 
<sip:ngcp-lb at x.x.x.142:5060;ngcpct=7369703a3132372e302e302e313a35303830>'

And on connect SPCE shows outbound peer IP (y.y.y.28) to customer and I 
believe that's completely wrong! They are not accessible from LAN and 
shows our topology

U 2015/04/08 16:22:39.604752 x.x.x.142:5060 -> 10.0.64.64:5060
SIP/2.0 200 OK'
Record-Route: 
<sip:127.0.0.1:5062;lr=on;ftag=668446~27154efa-6325-45a2-9e47-67e5d9302ebc-267715878;did=cfb.ccc2;mpd=ii;ice_caller=strip;ice_callee=strip;vsf=bWRoWFdrc29zTkN2Y1kzdHk0UmM->'
Record-Route: 
<sip:127.0.0.1;r2=on;lr=on;ftag=668446~27154efa-6325-45a2-9e47-67e5d9302ebc-267715878;nat=yes;ngcplb=yes;socket=udp:x.x.x.142:5060>'
Record-Route: 
<sip:x.x.x.142;r2=on;lr=on;ftag=668446~27154efa-6325-45a2-9e47-67e5d9302ebc-267715878;nat=yes;ngcplb=yes;socket=udp:x.x.x.142:5060>'
Via: SIP/2.0/UDP 10.0.64.64:5060;rport=5060;branch=z9hG4bK41e3371ccccc1'
From: 
<sip:12550 at 10.0.64.64>;tag=668446~27154efa-6325-45a2-9e47-67e5d9302ebc-267715878'
To: <sip:74996383868 at x.x.x.142>;tag=25F25514-55252B870008498F-D690C700'
Call-ID: 44882300-52512b87-2fae1-4040000a at 10.0.64.64'
CSeq: 101 INVITE'
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, 
SUBSCRIBE, NOTIFY, UPDATE, REGISTER'
Supported: replaces'
Content-Type: application/sdp'
Content-Length: 334'
Contact: 
<sip:ngcp-lb at x.x.x.142:5060;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31>'
'
v=0'
o=CiscoSystemsSIP-GW-UserAgent 4575 8817 IN IP4 y.y.y.28'
s=SIP Call'
c=IN IP4 y.y.y.28'
t=0 0'
m=audio 19144 RTP/AVP 18 4 8 0'
c=IN IP4 y.y.y.28'
a=rtpmap:18 G729/8000'
a=fmtp:18 annexb=yes'
a=rtpmap:4 G723/8000'
a=fmtp:4 bitrate=5.3;annexa=no'
a=rtpmap:8 PCMA/8000'
a=rtpmap:0 PCMU/8000'
a=sendrecv'
a=direction:both'


-- 
Serge S. Yuriev
Lead VoIP engineer