Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi, Thanks! do you think willl any other version will work? like version 3.0.x? Cheers, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
only trunk On Thu, Jul 7, 2011 at 2:17 AM, MingHon gming...@gmail.com wrote: Hi, Thanks! do you think willl any other version will work? like version 3.0.x? Cheers, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Alright thank you very much. i wil try to install it from trunk. Will report back later. :) Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi!! i finally able to replace the ip addr in the sdp body. but there is still some issue. will check tmr. anyway thank you very much. =) -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
hello List,
anyone could give some hints??
im still unable to rewrite the sdp body.
hope to hear from you all.
thanks
--
Regards,
MingHon
On Tue, Jul 5, 2011 at 3:49 PM, MingHon gming...@gmail.com wrote:
Hi List,
im facing an issue that my kamailio proxy did not replace the ip address in
the invite and 200OK sdp body.
my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user
my kamailio is listening on 192.168.1.3, also
define: advertised_address=175.136.223.112; advertised_port=5060;
and my asterisk is on 192.168.1.23.
sip signalling and rtp port forwarded to kamailio.
uacs from another nat register successfully.
if i put 2 lines of force_rtp_proxy(fcow,175.136.223.112);
i will get double ip addr in c and o but kamailio ignore my ip addr.
example i will get
c=IN IP4 192.168.1.3192.168.1.3
here is part of my simple script.
hope you can help.
thank you very much.
---cfg---
route[RTPPROXY] {
#!ifdef WITH_NAT
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy(fcow,175.136.223.112);
#force_rtp_proxy(fcow,175.136.223.112);
xlog(L_INFO,offer);
}
if (!has_totag()) add_rr_param(;nat=yes);
#!endif
return;
}
--
and here is the wireshark for uac INVITE and OK.
---INVITE-
ve0
EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0
Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080
Max-Forwards: 69
From: 101 sip:1...@aextddns.dyndns.info;tag=as032358a3
To: sip:102@192.168.1.3:5060
Contact: sip:102@192.168.1.23:5080
Call-ID: 416f6e09674ae9671bb7144a1cb11...@aextddns.dyndns.info
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18
Date: Tue, 05 Jul 2011 07:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 327
v=0
o=root 1639709788 1639709788 IN IP4 192.168.1.3
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.3
t=0 0
m=audio 10072 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
a=nortpproxy:yes
---200OK---
e90
ElE;pX4tSIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.200:5062
;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416
Record-Route: sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes
From: 101 sip:1...@aextddns.dyndns.info;tag=1796959074
To: sip:1...@aextddns.dyndns.info;tag=as2e4c0125
Call-ID: 1985782590@192.168.2.200
CSeq: 21 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.23:5080
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 403900934 403900934 IN IP4 192.168.1.23
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.23
t=0 0
m=audio 14420 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
My kamailio log.
---LOG--
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3
INFO: script: offer
-
double force_rtp_proxy
kamailio - asterisk [INVITE]-
Pyi-}E7V@:#pINVITE sip:1...@aextddns.dyndns.info SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=640933430;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0
Via: SIP/2.0/UDP 192.168.2.200:5062
;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648
From: 101 sip:1...@aextddns.dyndns.info;tag=640933430
To: sip:1...@aextddns.dyndns.info
Call-ID: 1909950509@192.168.2.200
CSeq: 21 INVITE
Contact: sip:101@175.138.21.31:2788
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 69
User-Agent: T20 9.41.0.80
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 334
v=0
o=20073 20073 IN IP4 192.168.1.3192.168.1.3
s=SDP data
c=IN IP4 192.168.1.3192.168.1.3
t=0 0
m=audio 1006410064 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=nortpproxy:yes
a=nortpproxy:yes
---LOG--
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi MingHon,
what do you want to achieve? If it is only about rewritibng the SDP,
then this will help you:
fix_nated_sdp(10, your-ip-here);
= 0x02 rewrite media IP address (c=) with the provided IP address
= 0x08 rewrite IP from origin description (o=) with the provided IP address
Kind regards,
Carsten
2011/7/6 MingHon gming...@gmail.com:
hello List,
anyone could give some hints??
im still unable to rewrite the sdp body.
hope to hear from you all.
thanks
--
Regards,
MingHon
On Tue, Jul 5, 2011 at 3:49 PM, MingHon gming...@gmail.com wrote:
Hi List,
im facing an issue that my kamailio proxy did not replace the ip address
in the invite and 200OK sdp body.
my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user
my kamailio is listening on 192.168.1.3, also
define: advertised_address=175.136.223.112; advertised_port=5060;
and my asterisk is on 192.168.1.23.
sip signalling and rtp port forwarded to kamailio.
uacs from another nat register successfully.
if i put 2 lines of force_rtp_proxy(fcow,175.136.223.112);
i will get double ip addr in c and o but kamailio ignore my ip addr.
example i will get
c=IN IP4 192.168.1.3192.168.1.3
here is part of my simple script.
hope you can help.
thank you very much.
---cfg---
route[RTPPROXY] {
#!ifdef WITH_NAT
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy(fcow,175.136.223.112);
#force_rtp_proxy(fcow,175.136.223.112);
xlog(L_INFO,offer);
}
if (!has_totag()) add_rr_param(;nat=yes);
#!endif
return;
}
--
and here is the wireshark for uac INVITE and OK.
---INVITE-
ve0
EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0
Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080
Max-Forwards: 69
From: 101 sip:1...@aextddns.dyndns.info;tag=as032358a3
To: sip:102@192.168.1.3:5060
Contact: sip:102@192.168.1.23:5080
Call-ID: 416f6e09674ae9671bb7144a1cb11...@aextddns.dyndns.info
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18
Date: Tue, 05 Jul 2011 07:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 327
v=0
o=root 1639709788 1639709788 IN IP4 192.168.1.3
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.3
t=0 0
m=audio 10072 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
a=nortpproxy:yes
---200OK---
e90
ElE;pX4tSIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.200:5062;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416
Record-Route: sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes
From: 101 sip:1...@aextddns.dyndns.info;tag=1796959074
To: sip:1...@aextddns.dyndns.info;tag=as2e4c0125
Call-ID: 1985782590@192.168.2.200
CSeq: 21 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.23:5080
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 403900934 403900934 IN IP4 192.168.1.23
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.23
t=0 0
m=audio 14420 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
My kamailio log.
---LOG--
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3
INFO: script: offer
-
double force_rtp_proxy
kamailio - asterisk [INVITE]-
Pyi-}E7V@:#pINVITE sip:1...@aextddns.dyndns.info SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=640933430;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0
Via: SIP/2.0/UDP
192.168.2.200:5062;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648
From: 101 sip:1...@aextddns.dyndns.info;tag=640933430
To: sip:1...@aextddns.dyndns.info
Call-ID: 1909950509@192.168.2.200
CSeq: 21 INVITE
Contact: sip:101@175.138.21.31:2788
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 69
User-Agent: T20 9.41.0.80
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 334
v=0
o=20073 20073 IN IP4 192.168.1.3192.168.1.3
s=SDP data
c=IN IP4 192.168.1.3192.168.1.3
t=0 0
m=audio 1006410064 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi MingHon,
can the RTPProxy actually use the External-Address? If i understood
you correctly you do NAT and port-forwarding between your Box and the
Internet on a firewall.
So, that is your problem. The RTPProxy tries to open an Port on the
External Address (which is not local) and fails. Thus he cannot relay
RTP-Traffic, and replies with an error to your kamailio box. If you
would move your RTPProxy to the world-wide-web, then you would not
have any problems. An RTPProxy behind NAT is horrible and you may run
into trouble.
Carsten
2011/7/6 MingHon gming...@gmail.com:
Hi Carsten,
i tried before putting external address but once i put external address i
will get error on my /var/log/messages
Jul 6 18:06:56 c5 /usr/local/sbin/kamailio[20025]: ERROR: rtpproxy
[rtpproxy.c:2211]: incorrect port 0 in reply from rtpproxy
Jul 6 18:06:56 c5 /usr/local/sbin/kamailio[20024]: ERROR: rtpproxy
[rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy
Jul 6 18:06:57 c5 /usr/local/sbin/kamailio[20025]: ERROR: rtpproxy
[rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy
both kamailio and rtpproxy is in one box and behind nat with one interface
ip address 192.168.1.3
in the router i already port forward 5060 udp and 1-2 udp to
192.168.1.3.
Please adv.
Thank you very much. :)
On Wed, Jul 6, 2011 at 6:01 PM, Carsten Bock cars...@ng-voice.com wrote:
Hi MingHon,
you should start your RTPProxy with -l external address (instead
of the internal address). The address provided will be used in the
signalling / replacement in the SDP.
Carsten
2011/7/6 MingHon gming...@gmail.com:
Hi,
Thanks for your reply..
RTPProxy and kamailio is running on the same centos box.
below is the command how i connect both RTPProxy and Kamailio
/Kamailio/
#!ifdef WITH_NAT
modparam(rtpproxy, rtpproxy_sock, udp:localhost:7722)
#!endif
/RTPProxy/
rtpproxy -l 192.168.1.3 -s udp:*:7722 -m 1 -M 2 -u user
/kamctl fifo nh_show_rtpp/
udp:localhost:7722:: set=0
index:: 0
disabled:: 0
weight:: 1
recheck_ticks:: 0
//
im using kamailio ver. 3.1.4 and rtpproxy ver. 1.2.1
Please advice..
Thank you very much for your help.
On Wed, Jul 6, 2011 at 4:16 PM, Carsten Bock cars...@ng-voice.com
wrote:
Hi,
Note:
The methods of rtpproxy-module will only replace the IP, if the
Kamailio can access the RTPProxy.
How is your RTPProxy connected to your Kamailio? Socket or TCP? Do you
have the Kamailio FIFO enabeld?
If you have the fifo enabled, you should check the following:
kamctl fifo nh_show_rtpp
You should see, that the Kamailio is connected to the RTPProxy. If no,
then that is your problem.
If the RTPProxy is connected and is listening on the TCP socket, then
you can do an ngrep to see the communication between Kamailio and
RTPProxy, which might help you further with your investigation.
Carsten
2011/7/6 MingHon gming...@gmail.com:
Hi Carsten,
no is not about just rewriting the SDP.
i need my UACs media to relay on my rtpproxy
currently my UACs are sending the media to a private ip.
my rtpproxy is in behind nat and UACs behind another nat.
On Wed, Jul 6, 2011 at 3:15 PM, Carsten Bock cars...@ng-voice.com
wrote:
Hi MingHon,
what do you want to achieve? If it is only about rewritibng the SDP,
then this will help you:
fix_nated_sdp(10, your-ip-here);
= 0x02 rewrite media IP address (c=) with the provided IP address
= 0x08 rewrite IP from origin description (o=) with the provided IP
address
Kind regards,
Carsten
2011/7/6 MingHon gming...@gmail.com:
hello List,
anyone could give some hints??
im still unable to rewrite the sdp body.
hope to hear from you all.
thanks
--
Regards,
MingHon
On Tue, Jul 5, 2011 at 3:49 PM, MingHon gming...@gmail.com
wrote:
Hi List,
im facing an issue that my kamailio proxy did not replace the ip
address
in the invite and 200OK sdp body.
my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user
my kamailio is listening on 192.168.1.3, also
define: advertised_address=175.136.223.112;
advertised_port=5060;
and my asterisk is on 192.168.1.23.
sip signalling and rtp port forwarded to kamailio.
uacs from another nat register successfully.
if i put 2 lines of force_rtp_proxy(fcow,175.136.223.112);
i will get double ip addr in c and o but kamailio ignore my ip
addr.
example i will get
c=IN IP4 192.168.1.3192.168.1.3
here is part of my simple script.
hope you can help.
thank you very much.
---cfg---
route[RTPPROXY] {
#!ifdef WITH_NAT
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy(fcow,175.136.223.112);
#force_rtp_proxy(fcow,175.136.223.112);
xlog(L_INFO,offer);
}
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi Carsten, I tried RTPProxy using external address. It work out from the box without any problem. UA successfully registered and RTP-Traffic relay work. Yup but now im trying to put the Box behind the nat. i refer to this thread http://lists.sip-router.org/pipermail/sr-users/2011-June/069223.html im able to put rtpproxy behind nat but need to modify the SDP body put in the proper ip address. so im trying to use force_rtp_proxy(co,external-address); to modify the (c=) and (o=) in SDP body. but in the SDP body i didnt see any changes of the ip address i tried adding several flags f,r,w but no luck. if i double my line force_rtp_proxy(co,external-address); in my SDP body i will get double private ip (c=192.168.1.3192.168.1.3) and (o=192.168.1.3192.168.1.3) thanks in advance, -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Can you try the trunk version? I remember fixing a bug related to the external IP, but I don't remember if it is available on a stable version. Regards, Ovidiu sas On Wed, Jul 6, 2011 at 11:16 AM, MingHon gming...@gmail.com wrote: Hi Carsten, I tried RTPProxy using external address. It work out from the box without any problem. UA successfully registered and RTP-Traffic relay work. Yup but now im trying to put the Box behind the nat. i refer to this thread http://lists.sip-router.org/pipermail/sr-users/2011-June/069223.html im able to put rtpproxy behind nat but need to modify the SDP body put in the proper ip address. so im trying to use force_rtp_proxy(co,external-address); to modify the (c=) and (o=) in SDP body. but in the SDP body i didnt see any changes of the ip address i tried adding several flags f,r,w but no luck. if i double my line force_rtp_proxy(co,external-address); in my SDP body i will get double private ip (c=192.168.1.3192.168.1.3) and (o=192.168.1.3192.168.1.3) thanks in advance, -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi Ovidiu, Thanks for the info. But why the command still available in the doc. http://www.kamailio.org/docs/modules/stable/modules_k/rtpproxy.html#id3024166 anyway i will try the trunk version. but may i know which version? is this ok? # svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5kamailio Thanks in advance. Cheers, MingHon On Thu, Jul 7, 2011 at 6:39 AM, Ovidiu Sas o...@voipembedded.com wrote: Can you try the trunk version? I remember fixing a bug related to the external IP, but I don't remember if it is available on a stable version. Regards, Ovidiu sas On Wed, Jul 6, 2011 at 11:16 AM, MingHon gming...@gmail.com wrote: Hi Carsten, I tried RTPProxy using external address. It work out from the box without any problem. UA successfully registered and RTP-Traffic relay work. Yup but now im trying to put the Box behind the nat. i refer to this thread http://lists.sip-router.org/pipermail/sr-users/2011-June/069223.html im able to put rtpproxy behind nat but need to modify the SDP body put in the proper ip address. so im trying to use force_rtp_proxy(co,external-address); to modify the (c=) and (o=) in SDP body. but in the SDP body i didnt see any changes of the ip address i tried adding several flags f,r,w but no luck. if i double my line force_rtp_proxy(co,external-address); in my SDP body i will get double private ip (c=192.168.1.3192.168.1.3) and (o=192.168.1.3192.168.1.3) thanks in advance, -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
You need to try the git trunk version. The command is available on all versions, but it doesn't work. Regards, Ovidiu Sas On Wed, Jul 6, 2011 at 10:01 PM, MingHon gming...@gmail.com wrote: Hi Ovidiu, Thanks for the info. But why the command still available in the doc. http://www.kamailio.org/docs/modules/stable/modules_k/rtpproxy.html#id3024166 anyway i will try the trunk version. but may i know which version? is this ok? # svn co http://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio Thanks in advance. Cheers, MingHon On Thu, Jul 7, 2011 at 6:39 AM, Ovidiu Sas o...@voipembedded.com wrote: Can you try the trunk version? I remember fixing a bug related to the external IP, but I don't remember if it is available on a stable version. Regards, Ovidiu sas On Wed, Jul 6, 2011 at 11:16 AM, MingHon gming...@gmail.com wrote: Hi Carsten, I tried RTPProxy using external address. It work out from the box without any problem. UA successfully registered and RTP-Traffic relay work. Yup but now im trying to put the Box behind the nat. i refer to this thread http://lists.sip-router.org/pipermail/sr-users/2011-June/069223.html im able to put rtpproxy behind nat but need to modify the SDP body put in the proper ip address. so im trying to use force_rtp_proxy(co,external-address); to modify the (c=) and (o=) in SDP body. but in the SDP body i didnt see any changes of the ip address i tried adding several flags f,r,w but no luck. if i double my line force_rtp_proxy(co,external-address); in my SDP body i will get double private ip (c=192.168.1.3192.168.1.3) and (o=192.168.1.3192.168.1.3) thanks in advance, -- Regards, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi, thanks.. may i know which version of kamailio? what the different between the stable tar ball and the git version? http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git is 3.1 okie? Cheers, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Try the latest 3.1. If it doesn't work, install from git. There should be instructions on the kamailio website aout how to do it. Regards, Ovidiu Sas On Thu, Jul 7, 2011 at 1:47 AM, MingHon gming...@gmail.com wrote: Hi, thanks.. may i know which version of kamailio? what the different between the stable tar ball and the git version? http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git is 3.1 okie? Cheers, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
3.1 was released on Oct 6, 2010 and my commit was on Oct 8, 2010: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commit;h=0af3944586c881f5d3ea0be19242cdf84906269e You will need to wait for 3.2 or install from trunk. Regards, Ovidiu Sas On Thu, Jul 7, 2011 at 1:49 AM, Ovidiu Sas o...@voipembedded.com wrote: Try the latest 3.1. If it doesn't work, install from git. There should be instructions on the kamailio website aout how to do it. Regards, Ovidiu Sas On Thu, Jul 7, 2011 at 1:47 AM, MingHon gming...@gmail.com wrote: Hi, thanks.. may i know which version of kamailio? what the different between the stable tar ball and the git version? http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git is 3.1 okie? Cheers, MingHon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
Hi List,
im facing an issue that my kamailio proxy did not replace the ip address in
the invite and 200OK sdp body.
my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user
my kamailio is listening on 192.168.1.3, also
define: advertised_address=175.136.223.112; advertised_port=5060;
and my asterisk is on 192.168.1.23.
sip signalling and rtp port forwarded to kamailio.
uacs from another nat register successfully.
if i put 2 lines of force_rtp_proxy(fcow,175.136.223.112);
i will get double ip addr in c and o but kamailio ignore my ip addr. example
i will get
c=IN IP4 192.168.1.3192.168.1.3
here is part of my simple script.
hope you can help.
thank you very much.
---cfg---
route[RTPPROXY] {
#!ifdef WITH_NAT
if (is_method(BYE)) {
unforce_rtp_proxy();
} else if (is_method(INVITE)){
force_rtp_proxy(fcow,175.136.223.112);
#force_rtp_proxy(fcow,175.136.223.112);
xlog(L_INFO,offer);
}
if (!has_totag()) add_rr_param(;nat=yes);
#!endif
return;
}
--
and here is the wireshark for uac INVITE and OK.
---INVITE-
ve0
EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0
Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080
Max-Forwards: 69
From: 101 sip:1...@aextddns.dyndns.info;tag=as032358a3
To: sip:102@192.168.1.3:5060
Contact: sip:102@192.168.1.23:5080
Call-ID: 416f6e09674ae9671bb7144a1cb11...@aextddns.dyndns.info
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.18
Date: Tue, 05 Jul 2011 07:20:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 327
v=0
o=root 1639709788 1639709788 IN IP4 192.168.1.3
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.3
t=0 0
m=audio 10072 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
a=nortpproxy:yes
---200OK---
e90
ElE;pX4tSIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.200:5062
;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416
Record-Route: sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes
From: 101 sip:1...@aextddns.dyndns.info;tag=1796959074
To: sip:1...@aextddns.dyndns.info;tag=as2e4c0125
Call-ID: 1985782590@192.168.2.200
CSeq: 21 INVITE
Server: Asterisk PBX 1.6.2.18
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:102@192.168.1.23:5080
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 403900934 403900934 IN IP4 192.168.1.23
s=Asterisk PBX 1.6.2.18
c=IN IP4 192.168.1.23
t=0 0
m=audio 14420 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
My kamailio log.
---LOG--
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3
INFO: script: offer
-
double force_rtp_proxy
kamailio - asterisk [INVITE]-
Pyi-}E7V@:#pINVITE sip:1...@aextddns.dyndns.info SIP/2.0
Record-Route: sip:192.168.1.3;lr=on;ftag=640933430;nat=yes
Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0
Via: SIP/2.0/UDP 192.168.2.200:5062
;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648
From: 101 sip:1...@aextddns.dyndns.info;tag=640933430
To: sip:1...@aextddns.dyndns.info
Call-ID: 1909950509@192.168.2.200
CSeq: 21 INVITE
Contact: sip:101@175.138.21.31:2788
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 69
User-Agent: T20 9.41.0.80
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 334
v=0
o=20073 20073 IN IP4 192.168.1.3192.168.1.3
s=SDP data
c=IN IP4 192.168.1.3192.168.1.3
t=0 0
m=audio 1006410064 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=nortpproxy:yes
a=nortpproxy:yes
---LOG--
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3
DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type application/sdp found valid
DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3
INFO: script: offer
---LOG--
--
Regards,
MingHon
___
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