21 Feb
2007
21 Feb
'07
6:51 p.m.
Raviprakash,
Your SDP messages are using private IPs for the RTP stream which carries
the voice traffic. This causing the media stream to be sent to the wrong
IP address, thus no audio.
Have you read through and tried to follow the directions in these documents?
*
*/OpenSER & NAT/*
o
Run RTPproxy on a remote host
<http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>
o
http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy
<http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>
<http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>
o
OpenSER and RTPProxy
<http://voip-info.org/wiki/view/OpenSER+And+RTPProxy>
o
__
<http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>_http://voip-info.org/wiki/view/OpenSER+And+RTPProxy
<http://openser.org/dokuwiki/doku.php/nat:remote-rtpproxy>_
o
OpenSER and Mediaproxy
<http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy>
o
http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy
A lot of good documentation has been written up on the OpenSER
application which can be found at this website.
http://openser.org/dokuwiki/doku.php
raviprakash sunkara wrote:
Hello Users,
I posted so many mail to users but no one reply my issue please
help me
openSER proxy is mysipdomain.com , and its private_ ip is 192.168.2.60
and
SIP server and Proxy is also in Behind
UAC's are Behind the NATs
--------------------- INVITE ---------------
U 59.144.88.7:5060 -> 192.168.2.60:5060
INVITE sip:9002@mysipdomain.com;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKe2a540a8170eb12a.
From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537.
To: <sip:9002@mysipdomain.com;user=phone>.
Call-ID: 1685867393(a)192.168.1.2.
CSeq: 1 INVITE.
Contact: Indian-2 <sip:8002@192.168.1.2:5060;user=phone;transport=udp>.
User-Agent: Cisco ATA 188 v3.2.1 atasip (050616A).
Expires: 300.
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE.
Supported: 100rel,replaces.
Content-Length: 245.
Content-Type: application/sdp.
v=0.
*o=8002 14279 14279 IN IP4 _192.168.1.2._*_ _
s=ATA186 Call.
c=IN IP4 192.168.1.2.
t=0 0.
m=audio 16386 RTP/AVP 0 4 8 101.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:4 G723/8000/1.
a=rtpmap:8 PCMA/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 192.168.2.60:5060 -> 61.17.248.68:3186
INVITE sip:9002@192.168.2.7:5060;user=phone;transport=udp SIP/2.0.
Max-Forwards: 10.
Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
;branch=z9hG4bKbd027751c869b9ff.
From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537.
To: <sip:9002@mysipdomain.com;user=phone>.
Call-ID: 1685867393(a)192.168.1.2.
CSeq: 2 INVITE.
Contact: Indian-2 <sip:8002@59.144.88.7:5060;user=phone;transport=udp>.
User-Agent: Cisco ATA 188 v3.2.1 atasip (050616A).
Expires: 300.
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE.
Supported: 100rel,replaces.
Content-Length: 265.
Content-Type: application/sdp.
v=0.
o=8002 14329 14329 IN IP4 192.168.1.2.
s=ATA186 Call.
c=IN IP4 192.168.1.2.
t=0 0.
m=audio 16386 RTP/AVP 0 4 8 101.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:4 G723/8000/1.
a=rtpmap:8 PCMA/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=direction:active.
------------------- RINGING------------
U 61.17.248.68:3186 -> 192.168.2.60:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
;branch=z9hG4bKbd027751c869b9ff.
Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537.
To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912.
Call-ID: 1685867393(a)192.168.1.2.
CSeq: 2 INVITE.
Require: 100rel.
RSeq: 1.
Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>.
Server: Cisco ATA 188 v3.2.1 atasip (050616A).
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE.
Content-Length: 0.
.
#
U 192.168.2.60:5060 -> 59.144.88.7:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
;branch=z9hG4bKbd027751c869b9ff.
Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537.
To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912.
Call-ID: 1685867393(a)192.168.1.2.
CSeq: 2 INVITE.
Require: 100rel.
RSeq: 1.
Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>.
Server: Cisco ATA 188 v3.2.1 atasip (050616A).
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE.
Content-Length: 0.
U 61.17.248.68:3186 -> 192.168.2.60:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
;branch=z9hG4bKbd027751c869b9ff.
Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537.
To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912.
Call-ID: 1685867393(a)192.168.1.2.
CSeq: 2 INVITE.
Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>.
Server: Cisco ATA 188 v3.2.1 atasip (050616A).
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE.
Supported: replaces.
Content-Length: 193.
Content-Type: application/sdp.
.
v=0.
*o=9002 27865 27865 IN IP4 _192.168.2.7_*_. _
s=ATA186 Call.
c=IN IP4 192.168.2.7.
t=0 0.
m=audio 16386 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 192.168.2.60:5060 -> 59.144.88.7:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
;branch=z9hG4bKbd027751c869b9ff.
Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537.
To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912.
Call-ID: 1685867393(a)192.168.1.2.
CSeq: 2 INVITE.
Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>.
Server: Cisco ATA 188 v3.2.1 atasip (050616A).
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE.
Supported: replaces.
Content-Length: 193.
Content-Type: application/sdp.
.
v=0.
o=9002 27865 27865 IN IP4 192.168.2.7.
s=ATA186 Call.
c=IN IP4 192.168.2.7.
t=0 0.
m=audio 16386 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 61.17.248.68:3186 -> 192.168.2.60:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.2.60;branch=z9hG4bKc466.901bd403.0.
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
;branch=z9hG4bKbd027751c869b9ff.
Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537.
To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912.
Call-ID: 1685867393(a)192.168.1.2.
CSeq: 2 INVITE.
Contact: 9002 <sip:9002@192.168.2.7:5060;user=phone;transport=udp>.
Server: Cisco ATA 188 v3.2.1 atasip (050616A).
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE.
Supported: replaces.
Content-Length: 193.
Content-Type: application/sdp.
.
v=0.
o=9002 27865 27865 IN IP4 192.168.2.7.
s=ATA186 Call.
c=IN IP4 192.168.2.7.
t=0 0.
m=audio 16386 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
#
U 192.168.2.60:5060 -> 59.144.88.7:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.2:5060;rport=5060;received=59.144.88.7
;branch=z9hG4bKbd027751c869b9ff.
Record-Route: <sip:192.168.2.60;lr=on;ftag=4240982537>.
From: Indian-2 <sip:8002@mysipdomain.com;user=phone>;tag=4240982537.
To: <sip:9002@mysipdomain.com;user=phone>;tag=1332822912.
Call-ID: 1685867393(a)192.168.1.2.
CSeq: 2 INVITE.
Contact: 9002 <sip:9002@61.17.248.68:3186;user=phone;transport=udp>.
Server: Cisco ATA 188 v3.2.1 atasip (050616A).
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK,
UPDATE.
Supported: replaces.
Content-Length: 193.
Content-Type: application/sdp.
v=0.
o=9002 27865 27865 IN IP4 192.168.2.7.
s=ATA186 Call.
c=IN IP4 192.168.2.7.
t=0 0.
m=audio 16386 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
------------------------------------------------------------------------
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