5 Aug
2013
5 Aug
'13
7:12 p.m.
I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex.
1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with LAN IP
2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN
with PUB IP 1.1.1.2) I am configured Registration of UA on Kamailio DB, and
on Asterisk side create a static peers with Kamailio LAN ip (host=2.2.2.2).
RTP Proxy question.
/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s udp:127.0.0.1 12221
.... kamailio.cfg ...
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
Testing call:
Whe User 1-100 calling User 1-101, on Asterisk side I see:
-- Called SIP/1-100(a)sip1.somedomain.com.ua
-- SIP/sip1.somedomain.com.ua-000004cf is ringing
-- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
0x15bc370 -- Probation passed - setting RTP source
address to
1.1.1.1:50868
0x7f2b6044bd10 -- Probation passed - setting RTP
source address to
1.1.1.1:35082
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027109, ts 000160, len
000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037469, ts 000160, len
000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027110, ts 000320, len
000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037470, ts 000320, len
000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027111, ts 000480, len
000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037471, ts 000480, len
000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027112, ts 000640, len
000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037472, ts 000640, len
000160)
Voice transfers OK.
But why not Kamailio LAN ip I receiving on the Asterisk side with the same
LAN?
And Kamailio log grep:
skynet:~ # tail -f /var/log/messages | grep rtpproxy
2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 41958 1.1.1.1
2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 39876 1.1.1.1
2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]: a=nortpproxy:yes
My goal is using Asterisk boxes behind Kamailio with the same LAN or even
OpeVPN Lan2Lan, with Public IP on Kamailio WAN for users registration and
RTP routing. So is strange to my, why RTPproxy not rewrite source of RTP
traffic from PUBLIC Kamailio IP to LAN Kamailio IP when user A calls B via
Asterisk?
5 Aug
5 Aug
11:02 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Hello,
you have to use rtpproxy in bridge mode, to route packets between the
two local network interfaces. There are many examples out there, one
shows even how to bridge between ipv4 and ipv4 networks -- you can use
it as reference:
- http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
Cheers,
Daniel
On 8/5/13 7:12 PM, Alexandr Usov wrote:
I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex.
1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with
LAN IP 2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1
pfsense LAN with PUB IP 1.1.1.2) I am configured Registration of UA on
Kamailio DB, and on Asterisk side create a static peers with Kamailio
LAN ip (host=2.2.2.2).
RTP Proxy question.
/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s udp:127.0.0.1 12221
.... kamailio.cfg ...
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
Testing call:
Whe User 1-100 calling User 1-101, on Asterisk side I see:
-- Called SIP/1-100(a)sip1.somedomain.com.ua
<mailto:1-100@sip1.somedomain.com.ua>
-- SIP/sip1.somedomain.com.ua-000004cf is ringing
-- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
0x15bc370 -- Probation passed - setting RTP
source address to
1.1.1.1:50868 <http://1.1.1.1:50868>
0x7f2b6044bd10 -- Probation passed - setting RTP
source
address to 1.1.1.1:35082 <http://1.1.1.1:35082>
Got RTP packet from 1.1.1.1:50868 <http://1.1.1.1:50868> (type 00,
seq 027109, ts 000160, len 000160)
Sent RTP packet to 1.1.1.1:35082 <http://1.1.1.1:35082> (type 00, seq
037469, ts 000160, len 000160)
Got RTP packet from 1.1.1.1:50868 <http://1.1.1.1:50868> (type 00,
seq 027110, ts 000320, len 000160)
Sent RTP packet to 1.1.1.1:35082 <http://1.1.1.1:35082> (type 00, seq
037470, ts 000320, len 000160)
Got RTP packet from 1.1.1.1:50868 <http://1.1.1.1:50868> (type 00,
seq 027111, ts 000480, len 000160)
Sent RTP packet to 1.1.1.1:35082 <http://1.1.1.1:35082> (type 00, seq
037471, ts 000480, len 000160)
Got RTP packet from 1.1.1.1:50868 <http://1.1.1.1:50868> (type 00,
seq 027112, ts 000640, len 000160)
Sent RTP packet to 1.1.1.1:35082 <http://1.1.1.1:35082> (type 00, seq
037472, ts 000640, len 000160)
Voice transfers OK.
But why not Kamailio LAN ip I receiving on the Asterisk side with the
same LAN?
And Kamailio log grep:
skynet:~ # tail -f /var/log/messages | grep rtpproxy
2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]: 3(25481)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]: 3(25481)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply:
63566 1.1.1.1
2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply:
41958 1.1.1.1
2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]: 4(25482)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]: 4(25482)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply:
39876 1.1.1.1
2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply:
43500 1.1.1.1
2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]: 4(25482)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]: 4(25482)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply:
63566 1.1.1.1
2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply:
43500 1.1.1.1
2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]: a=nortpproxy:yes
My goal is using Asterisk boxes behind Kamailio with the same LAN or
even OpeVPN Lan2Lan, with Public IP on Kamailio WAN for users
registration and RTP routing. So is strange to my, why RTPproxy not
rewrite source of RTP traffic from PUBLIC Kamailio IP to LAN Kamailio
IP when user A calls B via Asterisk?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
6 Aug
6 Aug
11:02 a.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Thank you for response!
A little difficult for me to find the same logic in my case with tutorial
of ipv4/ipv6 bridgin...
When I started
/usr/sbin/rtpproxy -u kamailio -l 1.1.1.1/2.2.2.2 -s udp:127.0.0.1 12221
There is no sound.
Is this a major to connect via unix sock?:
modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy/rtpproxy.sock")
2013/8/6 Daniel-Constantin Mierla <miconda(a)gmail.com>
Hello,
you have to use rtpproxy in bridge mode, to route packets between the two
local network interfaces. There are many examples out there, one shows even
how to bridge between ipv4 and ipv4 networks -- you can use it as reference:
- http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
Cheers,
Daniel
On 8/5/13 7:12 PM, Alexandr Usov wrote:
I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex.
1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with LAN IP
2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN
with PUB IP 1.1.1.2) I am configured Registration of UA on Kamailio DB, and
on Asterisk side create a static peers with Kamailio LAN ip (host=2.2.2.2).
RTP Proxy question.
/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s udp:127.0.0.1 12221
.... kamailio.cfg ...
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
Testing call:
Whe User 1-100 calling User 1-101, on Asterisk side I see:
-- Called SIP/1-100(a)sip1.somedomain.com.ua
-- SIP/sip1.somedomain.com.ua-000004cf is ringing
-- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
0x15bc370 -- Probation passed - setting RTP
source address to
1.1.1.1:50868
0x7f2b6044bd10 -- Probation passed - setting RTP
source address
to 1.1.1.1:35082
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027109, ts 000160,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037469, ts 000160,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027110, ts 000320,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037470, ts 000320,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027111, ts 000480,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037471, ts 000480,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027112, ts 000640,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037472, ts 000640,
len 000160)
Voice transfers OK.
But why not Kamailio LAN ip I receiving on the Asterisk side with the same
LAN?
And Kamailio log grep:
skynet:~ # tail -f /var/log/messages | grep rtpproxy
2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 41958 1.1.1.1
2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 39876 1.1.1.1
2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]: a=nortpproxy:yes
My goal is using Asterisk boxes behind Kamailio with the same LAN or
even OpeVPN Lan2Lan, with Public IP on Kamailio WAN for users registration
and RTP routing. So is strange to my, why RTPproxy not rewrite source of
RTP traffic from PUBLIC Kamailio IP to LAN Kamailio IP when user A calls B
via Asterisk?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
11:14 a.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Dear Alexandr,
You can connect Kamailio to RTPproxy via socket as well, use modparam like
this:
modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:12221")
Then if your rtprpoxy is started in bridged mode you should use the "i" and
"e" flags while you call the rtpproxy-manage() function in the kamailio.cfg
file.
The placement of both the flags sets the SDP c= param , so if you use "ie"
combination of flag then that is not equal to "ei" combination of the flag.
I also suggest that you turn on sip debug on the call receiving asterisk
and observe the SDP for an incoming call from Kamailio. that will help you
figure out the situation in SDP.
Best Regards,
Sammy
On Tue, Aug 6, 2013 at 2:02 AM, Alexandr Usov <blessendor(a)gmail.com> wrote:
Thank you for response!
A little difficult for me to find the same logic in my case with tutorial
of ipv4/ipv6 bridgin...
When I started
/usr/sbin/rtpproxy -u kamailio -l 1.1.1.1/2.2.2.2 -s udp:127.0.0.1 12221
There is no sound.
Is this a major to connect via unix sock?:
modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy/rtpproxy.sock")
2013/8/6 Daniel-Constantin Mierla <miconda(a)gmail.com>
Hello,
you have to use rtpproxy in bridge mode, to route packets between the two
local network interfaces. There are many examples out there, one shows even
how to bridge between ipv4 and ipv4 networks -- you can use it as reference:
- http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
Cheers,
Daniel
On 8/5/13 7:12 PM, Alexandr Usov wrote:
I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex.
1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with LAN IP
2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN
with PUB IP 1.1.1.2) I am configured Registration of UA on Kamailio DB, and
on Asterisk side create a static peers with Kamailio LAN ip (host=2.2.2.2).
RTP Proxy question.
/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s udp:127.0.0.1 12221
.... kamailio.cfg ...
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
Testing call:
Whe User 1-100 calling User 1-101, on Asterisk side I see:
-- Called SIP/1-100(a)sip1.somedomain.com.ua
-- SIP/sip1.somedomain.com.ua-000004cf is ringing
-- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
0x15bc370 -- Probation passed - setting RTP
source address to
1.1.1.1:50868
0x7f2b6044bd10 -- Probation passed - setting RTP
source address
to 1.1.1.1:35082
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027109, ts 000160,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037469, ts 000160,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027110, ts 000320,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037470, ts 000320,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027111, ts 000480,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037471, ts 000480,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027112, ts 000640,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037472, ts 000640,
len 000160)
Voice transfers OK.
But why not Kamailio LAN ip I receiving on the Asterisk side with the
same LAN?
And Kamailio log grep:
skynet:~ # tail -f /var/log/messages | grep rtpproxy
2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 41958 1.1.1.1
2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 39876 1.1.1.1
2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]: a=nortpproxy:yes
My goal is using Asterisk boxes behind Kamailio with the same LAN or
even OpeVPN Lan2Lan, with Public IP on Kamailio WAN for users registration
and RTP routing. So is strange to my, why RTPproxy not rewrite source of
RTP traffic from PUBLIC Kamailio IP to LAN Kamailio IP when user A calls B
via Asterisk?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
12:06 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
It seems I am undesrtand whereis problem can be found.
Original tutorial of Kamailio+Asterisk realtime integration (by Asipto)
containse settings for cheking if the "nat=yes" presents, but in Asterisk
11 I am using nat=force_rport,comedia.
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
BTW, we don't need NAT for asterisks peers, if we use Asterisk behinde
Kanailio LAN interface (2.2.2.2).
ToHost : 2.2.2.2
Addr->IP : 2.2.2.2:5060
If I cahnge to nat=no in the NATMANAGE - RTP debug still showing from
1.1.1.1 (public) kamailio IP.
Rtpproxy started in the bridge mode.
set_destination: Parsing <sip:2.2.2.2;r2=on;lr=on;nat=no> for address/port
to send to
set_destination: set destination to 2.2.2.2:5060
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
BYE sip:1-101@PUBLIC.CLIENT.PEER.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK1597848e
Route: <sip:2.2.2.2;r2=on;lr=on;nat=no>,<sip:1.1.1.1;r2=on;lr=on;nat=no>
Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003837, ts 016800, len
000160)
Got RTP packet from 2.2.2.2:42346 (type 00, seq 000986, ts 016960, len
000160)
Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003838, ts 016960, len
000160)
Got RTP packet from 2.2.2.2:42346 (type 00, seq 000987, ts 017120, len
000160)
Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003839, ts 017120, len
000160)
But no voicing)
2013/8/6 SamyGo <govoiper(a)gmail.com>
Dear Alexandr,
You can connect Kamailio to RTPproxy via socket as well, use modparam like
this:
modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:12221")
Then if your rtprpoxy is started in bridged mode you should use the "i"
and "e" flags while you call the rtpproxy-manage() function in the
kamailio.cfg file.
The placement of both the flags sets the SDP c= param , so if you use "ie"
combination of flag then that is not equal to "ei" combination of the flag.
I also suggest that you turn on sip debug on the call receiving asterisk
and observe the SDP for an incoming call from Kamailio. that will help you
figure out the situation in SDP.
Best Regards,
Sammy
On Tue, Aug 6, 2013 at 2:02 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
Thank you for response!
A little difficult for me to find the same logic in my case with tutorial
of ipv4/ipv6 bridgin...
When I started
/usr/sbin/rtpproxy -u kamailio -l 1.1.1.1/2.2.2.2 -s udp:127.0.0.1 12221
There is no sound.
Is this a major to connect via unix sock?:
modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy/rtpproxy.sock")
2013/8/6 Daniel-Constantin Mierla <miconda(a)gmail.com>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
you have to use rtpproxy in bridge mode, to route packets between the
two local network interfaces. There are many examples out there, one shows
even how to bridge between ipv4 and ipv4 networks -- you can use it as
reference:
- http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
Cheers,
Daniel
On 8/5/13 7:12 PM, Alexandr Usov wrote:
I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex.
1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with LAN IP
2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN
with PUB IP 1.1.1.2) I am configured Registration of UA on Kamailio DB, and
on Asterisk side create a static peers with Kamailio LAN ip (host=2.2.2.2).
RTP Proxy question.
/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s udp:127.0.0.1 12221
.... kamailio.cfg ...
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
Testing call:
Whe User 1-100 calling User 1-101, on Asterisk side I see:
-- Called SIP/1-100(a)sip1.somedomain.com.ua
-- SIP/sip1.somedomain.com.ua-000004cf is ringing
-- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
0x15bc370 -- Probation passed - setting RTP
source address to
1.1.1.1:50868
0x7f2b6044bd10 -- Probation passed - setting RTP
source address
to 1.1.1.1:35082
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027109, ts 000160,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037469, ts 000160,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027110, ts 000320,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037470, ts 000320,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027111, ts 000480,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037471, ts 000480,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027112, ts 000640,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037472, ts 000640,
len 000160)
Voice transfers OK.
But why not Kamailio LAN ip I receiving on the Asterisk side with the
same LAN?
And Kamailio log grep:
skynet:~ # tail -f /var/log/messages | grep rtpproxy
2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]: 3(25481) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 41958 1.1.1.1
2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 39876 1.1.1.1
2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]: a=nortpproxy:yes
2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]: 4(25482) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566 1.1.1.1
2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]: 6(25484) DEBUG:
rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500 1.1.1.1
2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]: a=nortpproxy:yes
My goal is using Asterisk boxes behind Kamailio with the same LAN or
even OpeVPN Lan2Lan, with Public IP on Kamailio WAN for users registration
and RTP routing. So is strange to my, why RTPproxy not rewrite source of
RTP traffic from PUBLIC Kamailio IP to LAN Kamailio IP when user A calls B
via Asterisk?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
12:13 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
You should not change the kamailio.cfg for nat=yes param, that works the
way it is. Yes you're right changing the NAT param in asterisk won't change
anything.
Please enable sip debug on asterisk and paste the complete INVITE/200OK
packets for the established call with no audio.
--
Sammy
On Tue, Aug 6, 2013 at 3:06 AM, Alexandr Usov <blessendor(a)gmail.com> wrote:
It seems I am undesrtand whereis problem can be
found.
Original tutorial of Kamailio+Asterisk realtime integration (by Asipto)
containse settings for cheking if the "nat=yes" presents, but in Asterisk
11 I am using nat=force_rport,comedia.
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
BTW, we don't need NAT for asterisks peers, if we use Asterisk behinde
Kanailio LAN interface (2.2.2.2).
ToHost : 2.2.2.2
Addr->IP : 2.2.2.2:5060
If I cahnge to nat=no in the NATMANAGE - RTP debug still showing from
1.1.1.1 (public) kamailio IP.
Rtpproxy started in the bridge mode.
set_destination: Parsing <sip:2.2.2.2;r2=on;lr=on;nat=no> for address/port
to send to
set_destination: set destination to 2.2.2.2:5060
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
BYE sip:1-101@PUBLIC.CLIENT.PEER.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK1597848e
Route: <sip:2.2.2.2;r2=on;lr=on;nat=no>,<sip:1.1.1.1;r2=on;lr=on;nat=no>
Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003837, ts 016800,
len 000160)
Got RTP packet from 2.2.2.2:42346 (type 00, seq 000986, ts 016960,
len 000160)
Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003838, ts 016960,
len 000160)
Got RTP packet from 2.2.2.2:42346 (type 00, seq 000987, ts 017120,
len 000160)
Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003839, ts 017120,
len 000160)
But no voicing)
2013/8/6 SamyGo <govoiper(a)gmail.com>
Dear Alexandr,
You can connect Kamailio to RTPproxy via socket as well, use modparam
like this:
modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:12221")
Then if your rtprpoxy is started in bridged mode you should use the "i"
and "e" flags while you call the rtpproxy-manage() function in the
kamailio.cfg file.
The placement of both the flags sets the SDP c= param , so if you use
"ie" combination of flag then that is not equal to "ei" combination
of the
flag.
I also suggest that you turn on sip debug on the call receiving asterisk
and observe the SDP for an incoming call from Kamailio. that will help you
figure out the situation in SDP.
Best Regards,
Sammy
On Tue, Aug 6, 2013 at 2:02 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Thank you for response!
A little difficult for me to find the same logic in my case with
tutorial of ipv4/ipv6 bridgin...
When I started
/usr/sbin/rtpproxy -u kamailio -l 1.1.1.1/2.2.2.2 -s udp:127.0.0.1 12221
There is no sound.
Is this a major to connect via unix sock?:
modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy/rtpproxy.sock")
2013/8/6 Daniel-Constantin Mierla <miconda(a)gmail.com>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
you have to use rtpproxy in bridge mode, to route packets between the
two local network interfaces. There are many examples out there, one shows
even how to bridge between ipv4 and ipv4 networks -- you can use it as
reference:
- http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6
Cheers,
Daniel
On 8/5/13 7:12 PM, Alexandr Usov wrote:
I have Kamailio on OpenSUSE with static real Public IP (WAN), for ex.
1.1.1.1. I have LAN IP 2.2.2.2. Asterisk as KVM virtual machine with LAN IP
2.2.2.101 and default GW not the SuSe (2.2.2.2), but 2.2.2.1 pfsense LAN
with PUB IP 1.1.1.2) I am configured Registration of UA on Kamailio DB, and
on Asterisk side create a static peers with Kamailio LAN ip (host=2.2.2.2).
RTP Proxy question.
/usr/sbin/rtpproxy -u daemon -l 1.1.1.1 -s udp:127.0.0.1 12221
.... kamailio.cfg ...
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
Testing call:
Whe User 1-100 calling User 1-101, on Asterisk side I see:
-- Called SIP/1-100(a)sip1.somedomain.com.ua
-- SIP/sip1.somedomain.com.ua-000004cf is ringing
-- SIP/sip1.somedomain.com.ua-000004cf answered SIP/1-101-000004ce
> 0x15bc370 -- Probation passed - setting RTP source address to
1.1.1.1:50868
> 0x7f2b6044bd10 -- Probation passed - setting RTP source
address to 1.1.1.1:35082
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027109, ts 000160,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037469, ts 000160,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027110, ts 000320,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037470, ts 000320,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027111, ts 000480,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037471, ts 000480,
len 000160)
Got RTP packet from 1.1.1.1:50868 (type 00, seq 027112, ts 000640,
len 000160)
Sent RTP packet to 1.1.1.1:35082 (type 00, seq 037472, ts 000640,
len 000160)
Voice transfers OK.
But why not Kamailio LAN ip I receiving on the Asterisk side with the
same LAN?
And Kamailio log grep:
skynet:~ # tail -f /var/log/messages | grep rtpproxy
2013-08-05T19:18:17.508760+03:00 skynet kamailio[25462]: 3(25481)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.508875+03:00 skynet kamailio[25462]: 3(25481)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566
1.1.1.1
2013-08-05T19:18:17.530765+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:17.530876+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 41958
1.1.1.1
2013-08-05T19:18:18.625815+03:00 skynet kamailio[25462]: 4(25482)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.627131+03:00 skynet kamailio[25462]: 4(25482)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 39876
1.1.1.1
2013-08-05T19:18:18.632649+03:00 skynet kamailio[25462]:
a=nortpproxy:yes
2013-08-05T19:18:18.648075+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.649615+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500
1.1.1.1
2013-08-05T19:18:18.653948+03:00 skynet kamailio[25462]:
a=nortpproxy:yes
2013-08-05T19:18:18.688603+03:00 skynet kamailio[25462]: 4(25482)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.689762+03:00 skynet kamailio[25462]: 4(25482)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 63566
1.1.1.1
2013-08-05T19:18:18.701062+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type
<application/sdp> found valid
2013-08-05T19:18:18.701405+03:00 skynet kamailio[25462]: 6(25484)
DEBUG: rtpproxy [rtpproxy.c:2624]: force_rtp_proxy(): proxy reply: 43500
1.1.1.1
2013-08-05T19:18:18.705506+03:00 skynet kamailio[25462]:
a=nortpproxy:yes
My goal is using Asterisk boxes behind Kamailio with the same LAN or
even OpeVPN Lan2Lan, with Public IP on Kamailio WAN for users registration
and RTP routing. So is strange to my, why RTPproxy not rewrite source of
RTP traffic from PUBLIC Kamailio IP to LAN Kamailio IP when user A calls B
via Asterisk?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
12:22 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
<------------>
Dial (.......) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19614
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@2.2.2.101:5080>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Tue, 06 Aug 2013 10:18:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 1885227245 1885227245 IN IP4 2.2.2.101
s=Asterisk
c=IN IP4 2.2.2.101
t=0 0
m=audio 19614 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-0000050f is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
12:31 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Hi again,
Still Missing 200OK for this call. It'll be helpful to send a complete
trace for the call coming in to the Asterisk at first place and then
Dialing out to the B-leg whose trace which you've just shared.
On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov <blessendor(a)gmail.com> wrote:
<------------>
Dial (.......) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19614
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@2.2.2.101:5080>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Tue, 06 Aug 2013 10:18:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 1885227245 1885227245 IN IP4 2.2.2.101
s=Asterisk
c=IN IP4 2.2.2.101
t=0 0
m=audio 19614 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-0000050f is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
12:36 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Sorry, It was call wothout answering.
I'm disable rtp debug and got full sip trace on asterisk side.
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 461
v=0
o=- 3584774018 3584774018 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:45033
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (18 headers 21 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Sending to 2.2.2.2:5060 (no NAT)
Using INVITE request as basis request - NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
Found peer '101' for '101' from 2.2.2.2:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 104
Found RTP audio format 109
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found audio description format speex for ID 103
Found audio description format speex for ID 102
Found audio description format speex for ID 104
Found audio description format iLBC for ID 109
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing),
combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
Looking for 101 in 1-internal (domain sip1.domain.com.ua)
list_route: hop: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
list_route: hop: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
-- Executing [101@1-internal:1] Macro("SIP/101-00000510",
"1-internal,101,60,rTt,0637679232,30,vm,Broker,101") in new stack
-- Executing [s@macro-1-internal:1] NoOp("SIP/101-00000510", "")
in new
stack
-- Executing [s@macro-1-internal:2] Dial("SIP/101-00000510", "SIP/
101(a)sip1.domain.com.ua,60,rTt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 14084
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk Cloud PBX 1.0
Date: Tue, 06 Aug 2013 10:33:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 2136064201 2136064201 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 14084 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-00000511 is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 57312 RTP/AVP 0 101
a=rtcp:57313
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:57312
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
set_destination: Parsing <sip:2.2.2.2;lr;r2=on;nat=yes> for address/port to
send to
set_destination: set destination to 2.2.2.2:5060
Transmitting (no NAT) to 2.2.2.2:5060:
ACK sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK67e4a7a4
Route: <sip:2.2.2.2;lr;r2=on;nat=yes>,<sip:1.1.1.1;lr;r2=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
Contact: <sip:101@192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 ACK
User-Agent: Asterisk Cloud PBX 1.0
Content-Length: 0
---
-- SIP/sip1.domain.com.ua-00000511 answered SIP/101-00000510
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 294
v=0
o=root 794877266 794877266 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
0x7f2b60530ea0 -- Probation passed - setting RTP
source address to
1.1.1.1:57312
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjogmI976-nkPYObMh8FDEf-ji4fnFUiCU
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 0 101
a=rtcp:45033
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (15 headers 13 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 270
v=0
o=root 794877266 794877267 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
0x7f2b60141b30 -- Probation passed - setting RTP
source address to
1.1.1.1:45032
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjyTfyg.XujqDRFK4QXdHXiQAydv.OoY6i
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
BYE sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
Max-Forwards: 16
From: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 18603 BYE
User-Agent: Telephone 1.0.4
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Scheduling destruction of SIP dialog '
025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua' in 32000 ms (Method:
BYE)
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
From: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 18603 BYE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (macro-1-internal, s, 2) exited non-zero on
'SIP/101-00000510' in macro '1-internal'
== Spawn extension (1-internal, 101, 1) exited non-zero on
'SIP/101-00000510'
Scheduling destruction of SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' in
6400 ms (Method: ACK)
set_destination: Parsing <sip:2.2.2.2;r2=on;lr=on;nat=yes> for address/port
to send to
set_destination: set destination to 2.2.2.2:5060
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Retransmitting #1 (no NAT) to 2.2.2.2:5060:
BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK5c0a5754
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' Method: ACK
2013/8/6 SamyGo <govoiper(a)gmail.com>
Hi again,
Still Missing 200OK for this call. It'll be helpful to send a complete
trace for the call coming in to the Asterisk at first place and then
Dialing out to the B-leg whose trace which you've just shared.
On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
<------------>
Dial (.......) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19614
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@2.2.2.101:5080>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Tue, 06 Aug 2013 10:18:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 1885227245 1885227245 IN IP4 2.2.2.101
s=Asterisk
c=IN IP4 2.2.2.101
t=0 0
m=audio 19614 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-0000050f is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
12:41 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Note:
Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this
described network (I am missed to change before copy-pasting here).
2013/8/6 Alexandr Usov <blessendor(a)gmail.com>
Sorry, It was call wothout answering.
I'm disable rtp debug and got full sip trace on asterisk side.
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 461
v=0
o=- 3584774018 3584774018 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:45033
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (18 headers 21 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Sending to 2.2.2.2:5060 (no NAT)
Using INVITE request as basis request - NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
Found peer '101' for '101' from 2.2.2.2:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 104
Found RTP audio format 109
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found audio description format speex for ID 103
Found audio description format speex for ID 102
Found audio description format speex for ID 104
Found audio description format iLBC for ID 109
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing),
combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
Looking for 101 in 1-internal (domain sip1.domain.com.ua)
list_route: hop: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
list_route: hop: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
-- Executing [101@1-internal:1] Macro("SIP/101-00000510",
"1-internal,101,60,rTt,0637679232,30,vm,Broker,101") in new stack
-- Executing [s@macro-1-internal:1] NoOp("SIP/101-00000510", "")
in
new stack
-- Executing [s@macro-1-internal:2] Dial("SIP/101-00000510", "SIP/
101(a)sip1.domain.com.ua,60,rTt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 14084
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk Cloud PBX 1.0
Date: Tue, 06 Aug 2013 10:33:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 2136064201 2136064201 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 14084 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-00000511 is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 57312 RTP/AVP 0 101
a=rtcp:57313
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:57312
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
set_destination: Parsing <sip:2.2.2.2;lr;r2=on;nat=yes> for address/port
to send to
set_destination: set destination to 2.2.2.2:5060
Transmitting (no NAT) to 2.2.2.2:5060:
ACK sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK67e4a7a4
Route: <sip:2.2.2.2;lr;r2=on;nat=yes>,<sip:1.1.1.1;lr;r2=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
Contact: <sip:101@192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 ACK
User-Agent: Asterisk Cloud PBX 1.0
Content-Length: 0
---
-- SIP/sip1.domain.com.ua-00000511 answered SIP/101-00000510
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 294
v=0
o=root 794877266 794877266 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
0x7f2b60530ea0 -- Probation passed - setting RTP
source address
to 1.1.1.1:57312
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjogmI976-nkPYObMh8FDEf-ji4fnFUiCU
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 0 101
a=rtcp:45033
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (15 headers 13 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 270
v=0
o=root 794877266 794877267 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
0x7f2b60141b30 -- Probation passed - setting RTP
source address
to 1.1.1.1:45032
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjyTfyg.XujqDRFK4QXdHXiQAydv.OoY6i
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
BYE sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
Max-Forwards: 16
From: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 18603 BYE
User-Agent: Telephone 1.0.4
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Scheduling destruction of SIP dialog '
025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua' in 32000 ms (Method:
BYE)
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
From: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 18603 BYE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (macro-1-internal, s, 2) exited non-zero on
'SIP/101-00000510' in macro '1-internal'
== Spawn extension (1-internal, 101, 1) exited non-zero on
'SIP/101-00000510'
Scheduling destruction of SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' in
6400 ms (Method: ACK)
set_destination: Parsing <sip:2.2.2.2;r2=on;lr=on;nat=yes> for
address/port to send to
set_destination: set destination to 2.2.2.2:5060
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Retransmitting #1 (no NAT) to 2.2.2.2:5060:
BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK5c0a5754
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
CSeq:
102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' Method: ACK
2013/8/6 SamyGo <govoiper(a)gmail.com>
Hi again,
Still Missing 200OK for this call. It'll be helpful to send a complete
trace for the call coming in to the Asterisk at first place and then
Dialing out to the B-leg whose trace which you've just shared.
On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
<------------>
Dial (.......) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19614
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@2.2.2.101:5080>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Tue, 06 Aug 2013 10:18:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 1885227245 1885227245 IN IP4 2.2.2.101
s=Asterisk
c=IN IP4 2.2.2.101
t=0 0
m=audio 19614 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-0000050f is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
12:48 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
I wonder who this belongs to : c=IN IP4 192.168.144.101
Also your Kamailio just sends the c=IN IP4 1.1.1.1 for the very first
incoming call that tells me that RTP proxy function is either running on
1.1.1.1 only or if its in bridging mode then you're not using the right
flag combination to use the 2.2.2.2 IP.
Is your kamailio set to have "mhomed=yes"; just wanted to know.
---
Sammy
On Tue, Aug 6, 2013 at 3:36 AM, Alexandr Usov <blessendor(a)gmail.com> wrote:
Sorry, It was call wothout answering.
I'm disable rtp debug and got full sip trace on asterisk side.
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 461
v=0
o=- 3584774018 3584774018 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:45033
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (18 headers 21 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Sending to 2.2.2.2:5060 (no NAT)
Using INVITE request as basis request - NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
Found peer '101' for '101' from 2.2.2.2:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 104
Found RTP audio format 109
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found audio description format speex for ID 103
Found audio description format speex for ID 102
Found audio description format speex for ID 104
Found audio description format iLBC for ID 109
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing),
combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
Looking for 101 in 1-internal (domain sip1.domain.com.ua)
list_route: hop: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
list_route: hop: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
-- Executing [101@1-internal:1] Macro("SIP/101-00000510",
"1-internal,101,60,rTt,0637679232,30,vm,Broker,101") in new stack
-- Executing [s@macro-1-internal:1] NoOp("SIP/101-00000510", "")
in
new stack
-- Executing [s@macro-1-internal:2] Dial("SIP/101-00000510", "SIP/
101(a)sip1.domain.com.ua,60,rTt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 14084
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk Cloud PBX 1.0
Date: Tue, 06 Aug 2013 10:33:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 2136064201 2136064201 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 14084 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-00000511 is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 57312 RTP/AVP 0 101
a=rtcp:57313
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:57312
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
set_destination: Parsing <sip:2.2.2.2;lr;r2=on;nat=yes> for address/port
to send to
set_destination: set destination to 2.2.2.2:5060
Transmitting (no NAT) to 2.2.2.2:5060:
ACK sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK67e4a7a4
Route: <sip:2.2.2.2;lr;r2=on;nat=yes>,<sip:1.1.1.1;lr;r2=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
Contact: <sip:101@192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 ACK
User-Agent: Asterisk Cloud PBX 1.0
Content-Length: 0
---
-- SIP/sip1.domain.com.ua-00000511 answered SIP/101-00000510
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 294
v=0
o=root 794877266 794877266 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
0x7f2b60530ea0 -- Probation passed - setting RTP
source address
to 1.1.1.1:57312
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjogmI976-nkPYObMh8FDEf-ji4fnFUiCU
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 0 101
a=rtcp:45033
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (15 headers 13 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 270
v=0
o=root 794877266 794877267 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
0x7f2b60141b30 -- Probation passed - setting RTP
source address
to 1.1.1.1:45032
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjyTfyg.XujqDRFK4QXdHXiQAydv.OoY6i
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
BYE sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
Max-Forwards: 16
From: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 18603 BYE
User-Agent: Telephone 1.0.4
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Scheduling destruction of SIP dialog '
025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua' in 32000 ms (Method:
BYE)
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
From: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 18603 BYE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (macro-1-internal, s, 2) exited non-zero on
'SIP/101-00000510' in macro '1-internal'
== Spawn extension (1-internal, 101, 1) exited non-zero on
'SIP/101-00000510'
Scheduling destruction of SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' in
6400 ms (Method: ACK)
set_destination: Parsing <sip:2.2.2.2;r2=on;lr=on;nat=yes> for
address/port to send to
set_destination: set destination to 2.2.2.2:5060
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Retransmitting #1 (no NAT) to 2.2.2.2:5060:
BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK5c0a5754
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
CSeq:
102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' Method: ACK
2013/8/6 SamyGo <govoiper(a)gmail.com>
Hi again,
Still Missing 200OK for this call. It'll be helpful to send a complete
trace for the call coming in to the Asterisk at first place and then
Dialing out to the B-leg whose trace which you've just shared.
On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
_______________________________________________
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
<------------>
Dial (.......) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19614
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@2.2.2.101:5080>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Tue, 06 Aug 2013 10:18:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 1885227245 1885227245 IN IP4 2.2.2.101
s=Asterisk
c=IN IP4 2.2.2.101
t=0 0
m=audio 19614 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-0000050f is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
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2:50 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
As I posted resently:
Note:
Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this
described network (I am missed to change before copy-pasting here).
2013/8/6 SamyGo <govoiper(a)gmail.com>
I wonder who this belongs to : c=IN IP4
192.168.144.101
Also your Kamailio just sends the c=IN IP4 1.1.1.1 for the very first
incoming call that tells me that RTP proxy function is either running on
1.1.1.1 only or if its in bridging mode then you're not using the right
flag combination to use the 2.2.2.2 IP.
Is your kamailio set to have "mhomed=yes"; just wanted to know.
---
Sammy
On Tue, Aug 6, 2013 at 3:36 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
Sorry, It was call wothout answering.
I'm disable rtp debug and got full sip trace on asterisk side.
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 461
v=0
o=- 3584774018 3584774018 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:45033
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (18 headers 21 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Sending to 2.2.2.2:5060 (no NAT)
Using INVITE request as basis request - NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
Found peer '101' for '101' from 2.2.2.2:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 104
Found RTP audio format 109
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found audio description format speex for ID 103
Found audio description format speex for ID 102
Found audio description format speex for ID 104
Found audio description format iLBC for ID 109
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing),
combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
Looking for 101 in 1-internal (domain sip1.domain.com.ua)
list_route: hop: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
list_route: hop: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
-- Executing [101@1-internal:1] Macro("SIP/101-00000510",
"1-internal,101,60,rTt,0637679232,30,vm,Broker,101") in new stack
-- Executing [s@macro-1-internal:1] NoOp("SIP/101-00000510", "")
in
new stack
-- Executing [s@macro-1-internal:2] Dial("SIP/101-00000510", "SIP/
101(a)sip1.domain.com.ua,60,rTt") in new stack
== Using SIP RTP CoS mark 5
Audio is at 14084
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk Cloud PBX 1.0
Date: Tue, 06 Aug 2013 10:33:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 2136064201 2136064201 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 14084 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK055f6241;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-00000511 is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK055f6241
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 57312 RTP/AVP 0 101
a=rtcp:57313
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:57312
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
set_destination: Parsing <sip:2.2.2.2;lr;r2=on;nat=yes> for address/port
to send to
set_destination: set destination to 2.2.2.2:5060
Transmitting (no NAT) to 2.2.2.2:5060:
ACK sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK67e4a7a4
Route: <sip:2.2.2.2;lr;r2=on;nat=yes>,<sip:1.1.1.1;lr;r2=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
To: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
Contact: <sip:101@192.168.144.101:5080>
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 102 ACK
User-Agent: Asterisk Cloud PBX 1.0
Content-Length: 0
---
-- SIP/sip1.domain.com.ua-00000511 answered SIP/101-00000510
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK671a.38ec3693.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPj3W1B1qUvS28C1WpfZg3jJ1JH1hp9zhM5
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 294
v=0
o=root 794877266 794877266 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
0x7f2b60530ea0 -- Probation passed - setting RTP
source address
to 1.1.1.1:57312
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjogmI976-nkPYObMh8FDEf-ji4fnFUiCU
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12894 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
INVITE sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 253
v=0
o=- 3584774018 3584774019 IN IP4 1.1.1.1
s=pjmedia
c=IN IP4 1.1.1.1
t=0 0
a=X-nat:0
m=audio 45032 RTP/AVP 0 101
a=rtcp:45033
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=nortpproxy:yes
<------------->
--- (15 headers 13 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.1.1.1:45032
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Length: 0
<------------>
Audio is at 18570
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK771a.cb41c8a3.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjQHAuHSkDhhq9S8uI-OhSHH8iKdVxBEP4
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 INVITE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@192.168.144.101:5080>
Content-Type: application/sdp
Require: timer
Content-Length: 270
v=0
o=root 794877266 794877267 IN IP4 192.168.144.101
s=Asterisk Cloud PBX 1.0
c=IN IP4 192.168.144.101
t=0 0
m=audio 18570 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
0x7f2b60141b30 -- Probation passed - setting RTP
source address
to 1.1.1.1:45032
<--- SIP read from UDP:2.2.2.2:5060 --->
ACK sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjyTfyg.XujqDRFK4QXdHXiQAydv.OoY6i
Max-Forwards: 16
From: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
To:
"Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 12895 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
BYE sip:101@192.168.144.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
Max-Forwards: 16
From: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 18603 BYE
User-Agent: Telephone 1.0.4
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 2.2.2.2:5060 (no NAT)
Scheduling destruction of SIP dialog '
025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua' in 32000 ms
(Method: BYE)
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKc32d.777a2532.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjHZ9R98C4pO9lP3OOvaTjVAe8rP7oPXnq
From: <sip:101@sip1.domain.com.ua>;tag=vNpq.SbQMuryrVC.Kitlti0D8L1sT.ZO
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as19dbc43e
Call-ID: 025be3512347f9f21b00d1930fa8f4bc(a)sip1.domain.com.ua
CSeq: 18603 BYE
Server: Asterisk Cloud PBX 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (macro-1-internal, s, 2) exited non-zero on
'SIP/101-00000510' in macro '1-internal'
== Spawn extension (1-internal, 101, 1) exited non-zero on
'SIP/101-00000510'
Scheduling destruction of SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii'
in 6400 ms (Method: ACK)
set_destination: Parsing <sip:2.2.2.2;r2=on;lr=on;nat=yes> for
address/port to send to
set_destination: set destination to 2.2.2.2:5060
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Retransmitting #1 (no NAT) to 2.2.2.2:5060:
BYE sip:101@CLIENT.GW.PUB.IP:17303;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.144.101:5080;branch=z9hG4bK5c0a5754
Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>,<sip:1.1.1.1;r2=on;lr=on;nat=yes>
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
CSeq: 102 BYE
User-Agent: Asterisk Cloud PBX 1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.144.101:5080;rport=5080;branch=z9hG4bK5c0a5754
Call-ID: NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as7db5bb42
To: "101" <sip:101@sip1.domain.com.ua
;tag=MM3JApujHR3kfZEZv.HkSj9rfUEssZDC
CSeq:
102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NlLTPqoVnhnqywTuVJ370D3TWdD0a8ii' Method:
ACK
2013/8/6 SamyGo <govoiper(a)gmail.com>
Hi again,
Still Missing 200OK for this call. It'll be helpful to send a complete
trace for the call coming in to the Asterisk at first place and then
Dialing out to the B-leg whose trace which you've just shared.
On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
<------------>
Dial (.......) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19614
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 2.2.2.2:5060:
INVITE sip:101@sip1.domain.com.ua SIP/2.0
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299
Max-Forwards: 70
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Contact: <sip:101@2.2.2.101:5080>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Tue, 06 Aug 2013 10:18:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 319
v=0
o=root 1885227245 1885227245 IN IP4 2.2.2.101
s=Asterisk
c=IN IP4 2.2.2.101
t=0 0
m=audio 19614 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIP/101(a)sip1.domain.com.ua
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
>;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 2.2.2.101:5080;branch=z9hG4bK3c47a299;rport=5080
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
CSeq: 102 INVITE
Server: kamailio (4.0.2 (x86_64/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.101:5080;rport=5080;branch=z9hG4bK3c47a299
Record-Route: <sip:1.1.1.1;lr;r2=on;nat=yes>
Record-Route: <sip:2.2.2.2;lr;r2=on;nat=yes>
Call-ID: 2ac37537499c919f01683582349522d6(a)sip1.domain.com.ua
From: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1b8070ba
To: <sip:101@sip1.domain.com.ua>;tag=JPsPQ1pCVKkH8Y1yAjFWmZk5c5PfyLNV
CSeq: 102 INVITE
Contact: "101" <sip:101@CLIENT.GW.PUB.IP:17303;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:2.2.2.2;lr;r2=on;nat=yes>
list_route: hop: <sip:1.1.1.1;lr;r2=on;nat=yes>
-- SIP/sip1.domain.com.ua-0000050f is ringing
<--- Transmitting (no NAT) to 2.2.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKe94e.aeadf975.0;received=2.2.2.2
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=CLIENT.GW.PUB.IP;rport=17303;branch=z9hG4bKPjP6ecaFom9elu9TnmL1KwJi0ktV5wYu5-
Record-Route: <sip:2.2.2.2;r2=on;lr=on;nat=yes>
Record-Route: <sip:1.1.1.1;r2=on;lr=on;nat=yes>
From: "101" <sip:101@sip1.domain.com.ua
>;tag=al9UV8T0rCNUe.DEp-8SXbeq3T4OiMxD
To: "Bomber" <sip:101@sip1.domain.com.ua>;tag=as1bd39f9d
Call-ID: YLXfZ6PZahPyigPKOKndbKf4QbJFw6U9
CSeq: 10050 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:101@2.2.2.101:5080>
Content-Length: 0
<------------>
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3:37 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
I wonder who this belongs to : c=IN IP4
192.168.144.101
This is Asterisk LAN IP (just nit changed by me before postin here to
2.2.2.101 for better reading WAN/LAN table).
Also your Kamailio just sends the c=IN IP4 1.1.1.1 for
the very first
incoming call that tells me that RTP proxy function is either running
on
1.1.1.1 only or if its in bridging mode then you're not using the right
flag combination to use the 2.2.2.2 IP.
Is your kamailio set to have "mhomed=yes";
just wanted to know.
I have (thinking yes=1):
mhomed=1
So, Kamailio have WAN interface (in my example 1.1.1.1 public IP), and LAN
- 2.2.2.2.
When I set "IE" on Kamailio in your Echo() test on Asterisk:
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage("IE");
I have RTP from LAN Kamailio IP - but no sounds:
Sent RTP packet to 2.2.2.2:47274 (type 00, seq 065027, ts 020800, len
000160)
Got RTP packet from 2.2.2.2:47274 (type 00, seq 008881, ts 020960, len
000160)
Sent RTP packet to 2.2.2.2:47274 (type 00, seq 065028, ts 020960, len
000160)
<--- SIP read from UDP:2.2.2.2:5060 --->
BYE sip:777@2.2.2.2.101:5080 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bKa28c.d8a404e3.0
Via: SIP/2.0/UDP 192.168.10.240:52396
;received=HIDDEN.PUB.CLIENT.IP;rport=38103;branch=z9hG4bKPjOZvyKasjc7sJHv2XbjndSqDEuDUSV2s-
Max-Forwards: 16
From: "1-101" <sip:1-101@sip1.domain.com.ua
;tag=rmbczI6t5gGLCfd3s7pVyH3L60RNJJ6Y
To:
<sip:777@sip1.domain.com.ua>;tag=as3da0acc7
Call-ID: 3PDsgSC91eH8McDiOJh.m03I-dcKf4Oc
CSeq: 17351 BYE
User-Agent: Telephone 1.0.4
Content-Length: 0
In "EI" combination - there are no RTP traffic on Asterisk side.
---
Sammy
4:40 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
I am gound flags for rtpproxy_manage(which help external (behind NAT) UA
registerd on Kamailio call to Echo() test extension on Asterisk - it is
"cwie". But for Peer-to-Peerm, registered on Kamailio and working though
Asterisk dialplan, it must be rtpproxy_manage("cwii").
Thanks to author of this example of kamailio.cfg
https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg
And from Kamailio documentation for rtpproxy_offer([flags [, ip_address]]):
-
*c* - flags to change the session-level SDP connection (c=) IP if
media-description also includes connection information.
-
*w* - flags that for the UA from which message is received, support
symmetric RTP must be forced.
Hope I am understand what I do =)
Thank you all for speaking with me! I'll be back... ASAP
4:55 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
My bad: "cwii" - sounds ok, but WAN Kamailio in RTP debug "from/to".
rtpproxy_manage("cwie"); - good for Echo() test,
when UA behind NAT, registered on Kamailip and calling Asterisk Echo() test
exten - voice perfect.
RTP - from/to only Kamailio LAN IP (2.2.2.2) - what is my goal.
But need to know, where I must put another flags for UA-t-UA calling, when
both behind teh NAT and registered with Kamailio?
Because I have WAN2LAN in RTP and both are IPs of Kamailio:
Sent RTP packet to 1.1.1.1:56324 (type 00, seq 003320, ts 006080, len
000160)
Got RTP packet from 2.2.2.2:62718 (type 00, seq 032129, ts 006240, len
000160)
Sent RTP packet to 1.1.1.1:56324 (type 00, seq 003321, ts 006240, len
000160)
Got RTP packet from 2.2.2.2:62718 (type 00, seq 032130, ts 006400, len
000160)
Sent RTP packet to 1.1.1.1:56324 (type 00, seq 003322, ts 006400, len
000160)
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage("cwie");
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
7 Aug
7 Aug
3:39 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Finally it's working:
#!define ASTERISK_LAN1 2.2.2.0/24
# End second LAN for PBXs - tunneling with LAN1 over OpenVPN:
#!define ASTERISK_LAN2 3.3.3.0.0/24
rtpproxy in bridge mode.
And route[NATMANAGE] from Asipto tutorial (
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb)
changed for my needs :
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
if((src_ip==ASTERISK_LAN1) || (src_ip==ASTERISK_LAN2))
{
rtpproxy_manage("cwei");
} else {
rtpproxy_manage("cwie");
}
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
Tested and worked on both tested Asterisk (LAN1 and LAN2). Kamailio just
one - in the LAN1.
6 Aug
6 Aug
12:55 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Note:
Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this
described network (I am missed to change before copy-pasting here).
12:58 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Please check the rtpproxy function and paste the way it is written in your
configuration file. Share the output of "ps -ef | grep rtpproxy" and
"netstat -pln|grep rtpproxy"
--
Sammy
On Tue, Aug 6, 2013 at 3:55 AM, Alexandr Usov <blessendor(a)gmail.com> wrote:
Note:
Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this
described network (I am missed to change before copy-pasting here).
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
1:06 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
ps -ef | grep rtpproxy
kamailio 15853 1 0 12:41 ? 00:00:01 /usr/sbin/rtpproxy -u
kamailio -l 1.1.1.1 2.2.2.2 -s unix:/var/run/rtpproxy.sock
netstat -pln|grep rtpproxy
unix 2 [ ACC ] STREAM LISTENING 238561268
15853/rtpproxy /var/run/rtpproxy.sock
I'm trying rtpproxy_manage("ei"); rtpproxy_manage("ie");
rtpproxy_manage();
- the same if rtpproxy in bridged mode.
Config cuts:
#!define WITH_NAT
...
#!ifdef WITH_NAT
# ----- rtpproxy params -----
# modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:12221")
modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from",
"sip:pinger@sip.myrealhostdomain.com")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp",
"$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
....
# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
exit;
}
}
...
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
2013/8/6 SamyGo <govoiper(a)gmail.com>
Please check the rtpproxy function and paste the way
it is written in your
configuration file. Share the output of "ps -ef | grep rtpproxy" and
"netstat -pln|grep rtpproxy"
--
Sammy
On Tue, Aug 6, 2013 at 3:55 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
Note:
Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this
described network (I am missed to change before copy-pasting here).
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
1:15 p.m.
New subject: (Kamailio/RTPproyx) Public -> LAN -> Asterisk (1, 2, .., n)
Ok thanks,
All fine except the rtpproxy_manage function is just out in the open and
since you're in bridging mode you need to realize that your kamailio may
receive calls from the WAN Interface or from the Asterisks on LAN
interface. So if you're to bridge the RTPs at the proxy then how can
Kamailio detect that which combination of flags to use in rtpproxy_manage()
function?
As an example if you replace your rtpproxy_manage(); with
rtpproxy_manage("ei"); and make a call from SIP user to 777 in asterisk
dialplan where 777 is echo()
exten => 777,1,Answer()
same => n,Echo()
Take a sip trace now and see that you get the desired LAN IP of kamailio
coming in on Asterisk ? revert the "ei" to "ie" if not.
This simple test should get you a two way audio established for any call
from Internet to the Asterisk via Kamailio.
The Next step is to figure out that your call is coming from Asterisk and
wants to go out from kamailio's WAN interface. use inverse flag combination
here and let the call go out.
I hope it works for you now.
--
Sammy
On Tue, Aug 6, 2013 at 4:06 AM, Alexandr Usov <blessendor(a)gmail.com> wrote:
ps -ef | grep rtpproxy
kamailio 15853 1 0 12:41 ? 00:00:01 /usr/sbin/rtpproxy -u
kamailio -l 1.1.1.1 2.2.2.2 -s unix:/var/run/rtpproxy.sock
netstat -pln|grep rtpproxy
unix 2 [ ACC ] STREAM LISTENING 238561268
15853/rtpproxy /var/run/rtpproxy.sock
I'm trying rtpproxy_manage("ei"); rtpproxy_manage("ie");
rtpproxy_manage(); - the same if rtpproxy in bridged mode.
Config cuts:
#!define WITH_NAT
...
#!ifdef WITH_NAT
# ----- rtpproxy params -----
# modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:12221")
modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from",
"sip:pinger@sip.myrealhostdomain.com
")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp",
"$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
....
# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
exit;
}
}
...
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
2013/8/6 SamyGo <govoiper(a)gmail.com>
Please check the rtpproxy function and paste the
way it is written in
your configuration file. Share the output of "ps -ef | grep rtpproxy" and
"netstat -pln|grep rtpproxy"
--
Sammy
On Tue, Aug 6, 2013 at 3:55 AM, Alexandr Usov <blessendor(a)gmail.com>wrote;wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Note:
Asrterisk LAN IP real 192.168.144.101 but must be 2.2.2.101 in this
described network (I am missed to change before copy-pasting here).
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3937
days inactive
3939
days old
19 comments
3 participants
participants (3)
-
Alexandr Usov -
Daniel-Constantin Mierla -
SamyGo