21 Feb
2012
21 Feb
'12
10:31 p.m.
Greetings,
I'm not sure if I found a bug, or if I just have something completely misconfigured...
I'm a total newb with Kamailio, working on a proof of concept design.
Here's my configuration:
provider -> nat firewall -> kamailio/rtpproxy -> asterisk
For outbound calls from a phone registered to asterisk via kamailio, I'm trying to use
fix_nated_sdp("2", "10.50.50.8") to rewrite the media ip address to
resolve my audio issues, where 10.50.50.8 is the address outside my firewall. What
I'm running into is the 'c=' line doesn't get re-written properly... it
inserts the specified address in front of the existing address, and I end up with the
following line in my INVITE:
c=IN IP4 10.50.50.810.0.10.10
I have the fix_nated_sdp command under route[sipout], because I only want to use it on
calls being sent outside the nat firewall.
Here's the sip invite without the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19165551212@xxx.xxx.xxx.xxx SIP/2.0
Record-Route: <sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060
Max-Forwards: 69
From: "1009" <sip:1009@10.0.10.11>;tag=as5498b77e
To: <sip:19165551212@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060>
Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:06:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309
P-hint: outbound
v=0
o=root 604360056 604360056 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.0.10.10
t=0 0
m=audio 9702 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=ptime:20
a=sendrecv
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Here's the sip invite with the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19167828326@xxx.xxx.xxx.xxx SIP/2.0
Record-Route: <sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060
Max-Forwards: 69
From: "1009" <sip:1009@10.0.10.11>;tag=as49e00c81
To: <sip:19167828326@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060>
Call-ID: 4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 347
P-hint: outbound
v=0
o=root 1009117068 1009117068 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.50.50.8.10.0.10.10
t=0 0
m=audio 13540 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=ptime:20
a=sendrecv
a=oldmediaip:10.0.10.11
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Is this a bug, or is it likely I have something else screwed up?
Thank you in advance for your assistance - this list is an incredible resource!
-Ric
22 Feb
22 Feb
3:51 a.m.
Hello,
can you set debug=3 in the config file and send the output (syslog
messages) of processing such invite?
Cheers,
Daniel
On 2/22/12 4:31 AM, Ric Marques wrote:
Greetings,
I'm not sure if I found a bug, or if I just have something completely
misconfigured... I'm a total newb with Kamailio, working on a proof of
concept design.
Here's my configuration:
provider -> nat firewall -> kamailio/rtpproxy -> asterisk
For outbound calls from a phone registered to asterisk via kamailio,
I'm trying to use fix_nated_sdp("2", "10.50.50.8") to rewrite the
media ip address to resolve my audio issues, where 10.50.50.8 is the
address outside my firewall. What I'm running into is the 'c=' line
doesn't get re-written properly... it inserts the specified address in
front of the existing address, and I end up with the following line in
my INVITE:
c=IN IP4 10.50.50.810.0.10.10
I have the fix_nated_sdp command under route[sipout], because I only
want to use it on calls being sent outside the nat firewall.
Here's the sip invite without the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19165551212@xxx.xxx.xxx.xxx SIP/2.0
Record-Route: <sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060
Max-Forwards: 69
From: "1009" <sip:1009@10.0.10.11>;tag=as5498b77e
To: <sip:19165551212@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060>
Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:06:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309
P-hint: outbound
v=0
o=root 604360056 604360056 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.0.10.10
t=0 0
m=audio 9702 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=ptime:20
a=sendrecv
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Here's the sip invite with the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19167828326@xxx.xxx.xxx.xxx SIP/2.0
Record-Route: <sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060
Max-Forwards: 69
From: "1009" <sip:1009@10.0.10.11>;tag=as49e00c81
To: <sip:19167828326@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060>
Call-ID: 4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 347
P-hint: outbound
v=0
o=root 1009117068 1009117068 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.50.50.8.10.0.10.10
t=0 0
m=audio 13540 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=ptime:20
a=sendrecv
a=oldmediaip:10.0.10.11
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Is this a bug, or is it likely I have something else screwed up?
Thank you in advance for your assistance - this list is an incredible
resource!
-Ric
_______________________________________________
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
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
1:50 p.m.
Daniel -
Thank you for your assistance..
first, here's the sections of my routing where I'm calling fix_nated_sdp, and
subsequent call:
# Routing to foreign domains
route[SIPOUT] {
xlog("---------------------- checking of outbound to somewhere else
-----------------------------------------");
if (!uri==myself)
{
xlog("<---------------------------------- Sending
call out to some other domain ------------------------------>");
append_hf("P-hint: outbound\r\n");
set_advertised_address("10.50.50.8");
xlog("--------------------------bing--------------------------");
fix_nated_sdp("2", "10.50.50.8");
xlog("--------------------------bong--------------------------");
route(RELAY);
}
}
route[RELAY] {
xlog("------------------------------ relaying
-------------------------------");
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|SUBSCRIBE")) {
t_on_branch("MANAGE_BRANCH");
t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
xlog("------------------------------ exiting relaying
-------------------------------");
exit;
}
and here's the section of the log where that's found:
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
---------------------- checking of outbound to somewhere else
-----------------------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[socket_info.c:502]: grep_sock_info - checking if host==us: 13==10 &&
[xxx.xxx.xxx.xxx] == [10.0.10.10]
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[socket_info.c:505]: grep_sock_info - checking if port 5060 matches port 5060
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core> [forward.c:448]:
check_self: host != me
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
<---------------------------------- Sending call out to some other domain
------------------------------>
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
--------------------------bing--------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: nathelper
[nhelpr_funcs.c:148]: type <application/sdp> found valid
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
--------------------------bong--------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
------------------------------ relaying -------------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_lookup.c:1379]: DEBUG:
t_newtran: msg id=3 , global msg id=3 , T on entrance=(nil)
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_lookup.c:527]:
t_lookup_request: start searching: hash=34053, isACK=0
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_lookup.c:485]: DEBUG:
RFC3261 transaction matching failed
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_lookup.c:709]: DEBUG:
t_lookup_request: no transaction found
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_hooks.c:374]: DBG:
trans=0x7fb56c0478e8, callback type 1, id 0 entered
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_funcs.c:351]: SER: new
INVITE
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[msg_translator.c:204]: check_via_address(10.0.10.11, 10.0.10.11, 0)
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_reply.c:667]: DEBUG:
reply sent out. buf=0x7fb571968880: SIP/2.0 100 trying -..., shmem=0x7fb56c049eb8: SIP/2.0
100 trying -
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_reply.c:677]: DEBUG:
_reply_light: finished
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <script>: new branch [1]
to sip:19165551212@xxx.xxx.xxx.xxx
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: siputils [checks.c:104]: no
totag
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[parser/sdp/sdp_helpr_funcs.c:479]: located IP address [10.0.10.11] in `o=' field
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[parser/sdp/sdp_helpr_funcs.c:479]: located IP address [10.0.10.11] in `c=' field
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: rtpproxy
[rtpproxy_funcs.c:148]: type <application/sdp> found valid
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: rtpproxy [rtpproxy.c:2237]:
proxy reply: 38946 10.0.10.10#012
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: siputils [checks.c:104]: no
totag
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[msg_translator.c:457]: clen_builder: content-length: 347 (347)
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[msg_translator.c:204]: check_via_address(10.0.10.11, 10.0.10.11, 0)
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm [t_funcs.c:388]: SER: new
transaction fwd'ed
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
------------------------------ exiting relaying -------------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core> [usr_avp.c:644]:
DEBUG:destroy_avp_list: destroying list (nil)
Walking through the log makes me think that because I'm using rtpproxy and nathelper,
when the t_relay fires it errantly appends the address for rtpproxy to the c= line...
Am I going about this all wrong - is there a better approach?
Ric
From: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
Sent: Wednesday, February 22, 2012 12:52 AM
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List
Cc: Ric Marques
Subject: Re: [SR-Users] fix_nated_sdp issue
Hello,
can you set debug=3 in the config file and send the output (syslog messages) of processing
such invite?
Cheers,
Daniel
On 2/22/12 4:31 AM, Ric Marques wrote:
Greetings,
I'm not sure if I found a bug, or if I just have something completely misconfigured...
I'm a total newb with Kamailio, working on a proof of concept design.
Here's my configuration:
provider -> nat firewall -> kamailio/rtpproxy -> asterisk
For outbound calls from a phone registered to asterisk via kamailio, I'm trying to use
fix_nated_sdp("2", "10.50.50.8") to rewrite the media ip address to
resolve my audio issues, where 10.50.50.8 is the address outside my firewall. What
I'm running into is the 'c=' line doesn't get re-written properly... it
inserts the specified address in front of the existing address, and I end up with the
following line in my INVITE:
c=IN IP4 10.50.50.810.0.10.10
I have the fix_nated_sdp command under route[sipout], because I only want to use it on
calls being sent outside the nat firewall.
Here's the sip invite without the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19165551212@xxx.xxx.xxx.xxx SIP/2.0
Record-Route:
<sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes><sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060
Max-Forwards: 69
From: "1009"
<sip:1009@10.0.10.11><sip:1009@10.0.10.11>;tag=as5498b77e
To: <sip:19165551212@xxx.xxx.xxx.xxx><sip:19165551212@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060><sip:1009@10.0.10.11:5060>
Call-ID:
06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060<mailto:06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060>
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:06:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309
P-hint: outbound
v=0
o=root 604360056 604360056 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.0.10.10
t=0 0
m=audio 9702 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=ptime:20
a=sendrecv
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Here's the sip invite with the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19167828326@xxx.xxx.xxx.xxx SIP/2.0
Record-Route:
<sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes><sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060
Max-Forwards: 69
From: "1009"
<sip:1009@10.0.10.11><sip:1009@10.0.10.11>;tag=as49e00c81
To: <sip:19167828326@xxx.xxx.xxx.xxx><sip:19167828326@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060><sip:1009@10.0.10.11:5060>
Call-ID:
4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060<mailto:4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060>
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 347
P-hint: outbound
v=0
o=root 1009117068 1009117068 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.50.50.8.10.0.10.10
t=0 0
m=audio 13540 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=ptime:20
a=sendrecv
a=oldmediaip:10.0.10.11
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Is this a bug, or is it likely I have something else screwed up?
Thank you in advance for your assistance - this list is an incredible resource!
-Ric
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
24 Feb
24 Feb
2:14 a.m.
Hello,
On 2/22/12 7:50 PM, Ric Marques wrote:
Daniel -
Thank you for your assistance..
first, here's the sections of my routing where I'm calling
fix_nated_sdp, and subsequent call:
# Routing to foreign domains
route[SIPOUT] {
xlog("---------------------- checking of outbound to
somewhere else -----------------------------------------");
if (!uri==myself)
{
xlog("<---------------------------------- Sending call out to some
other domain ------------------------------>");
append_hf("P-hint: outbound\r\n");
set_advertised_address("10.50.50.8");
xlog("--------------------------bing--------------------------");
fix_nated_sdp("2", "10.50.50.8");
xlog("--------------------------bong--------------------------");
route(RELAY);
}
}
route[RELAY] {
xlog("------------------------------ relaying
-------------------------------");
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|SUBSCRIBE")) {
t_on_branch("MANAGE_BRANCH");
t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
xlog("------------------------------ exiting relaying
-------------------------------");
exit;
}
and here's the section of the log where that's found:
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
---------------------- checking of outbound to somewhere else
-----------------------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[socket_info.c:502]: grep_sock_info - checking if host==us: 13==10 &&
[xxx.xxx.xxx.xxx] == [10.0.10.10]
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[socket_info.c:505]: grep_sock_info - checking if port 5060 matches
port 5060
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[forward.c:448]: check_self: host != me
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
<---------------------------------- Sending call out to some other
domain ------------------------------>
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
--------------------------bing--------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: nathelper
[nhelpr_funcs.c:148]: type <application/sdp> found valid
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
--------------------------bong--------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
------------------------------ relaying -------------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm
[t_lookup.c:1379]: DEBUG: t_newtran: msg id=3 , global msg id=3 , T on
entrance=(nil)
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm
[t_lookup.c:527]: t_lookup_request: start searching: hash=34053, isACK=0
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm
[t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm
[t_lookup.c:709]: DEBUG: t_lookup_request: no transaction found
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm
[t_hooks.c:374]: DBG: trans=0x7fb56c0478e8, callback type 1, id 0 entered
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm
[t_funcs.c:351]: SER: new INVITE
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[msg_translator.c:204]: check_via_address(10.0.10.11, 10.0.10.11, 0)
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[mem/shm_mem.c:111]: WARNING:vqm_resize: resize(0) called
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm
[t_reply.c:667]: DEBUG: reply sent out. buf=0x7fb571968880: SIP/2.0
100 trying -..., shmem=0x7fb56c049eb8: SIP/2.0 100 trying -
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm
[t_reply.c:677]: DEBUG: _reply_light: finished
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <script>:
new branch [1] to sip:19165551212@xxx.xxx.xxx.xxx
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: siputils
[checks.c:104]: no totag
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[parser/sdp/sdp_helpr_funcs.c:479]: located IP address [10.0.10.11] in
`o=' field
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[parser/sdp/sdp_helpr_funcs.c:479]: located IP address [10.0.10.11] in
`c=' field
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: rtpproxy
[rtpproxy_funcs.c:148]: type <application/sdp> found valid
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: rtpproxy
[rtpproxy.c:2237]: proxy reply: 38946 10.0.10.10#012
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: siputils
[checks.c:104]: no totag
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[msg_translator.c:457]: clen_builder: content-length: 347 (347)
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[msg_translator.c:204]: check_via_address(10.0.10.11, 10.0.10.11, 0)
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: tm
[t_funcs.c:388]: SER: new transaction fwd'ed
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: ERROR: <script>:
------------------------------ exiting relaying
-------------------------------
Feb 22 10:14:32 demo /usr/local/sbin/kamailio[3602]: DEBUG: <core>
[usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil)
Walking through the log makes me think that because I'm using rtpproxy
and nathelper, when the t_relay fires it errantly appends the address
for rtpproxy to the c= line...
Am I going about this all wrong - is there a better approach?
you cannot use manage_rtpproxy (or other functions from rtpproxy module
updating the sdp) with fix_nated_sdp() because of the way changes are
applied to the sip message. When using both, it results in concatenation
of the two IP -- it is why I asked about the log, expecting you used
such two functions.
You may try using msg_apply_changes() in between such two functions, but
I recommend making the config file decision in a way that you execute
only one such function. For example, you can set a flag after using such
functions and before using another one test that flag.
Cheers,
Daniel
Ric
*From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com]
*Sent:* Wednesday, February 22, 2012 12:52 AM
*To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
Users Mailing List
*Cc:* Ric Marques
*Subject:* Re: [SR-Users] fix_nated_sdp issue
Hello,
can you set debug=3 in the config file and send the output (syslog
messages) of processing such invite?
Cheers,
Daniel
On 2/22/12 4:31 AM, Ric Marques wrote:
Greetings,
I'm not sure if I found a bug, or if I just have something completely
misconfigured... I'm a total newb with Kamailio, working on a proof of
concept design.
Here's my configuration:
provider -> nat firewall -> kamailio/rtpproxy -> asterisk
For outbound calls from a phone registered to asterisk via kamailio,
I'm trying to use fix_nated_sdp("2", "10.50.50.8") to rewrite the
media ip address to resolve my audio issues, where 10.50.50.8 is the
address outside my firewall. What I'm running into is the 'c=' line
doesn't get re-written properly... it inserts the specified address in
front of the existing address, and I end up with the following line in
my INVITE:
c=IN IP4 10.50.50.810.0.10.10
I have the fix_nated_sdp command under route[sipout], because I only
want to use it on calls being sent outside the nat firewall.
Here's the sip invite without the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19165551212@xxx.xxx.xxx.xxx SIP/2.0
Record-Route: <sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060
Max-Forwards: 69
From: "1009" <sip:1009@10.0.10.11>;tag=as5498b77e
To: <sip:19165551212@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060>
Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060
<mailto:06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060>
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:06:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309
P-hint: outbound
v=0
o=root 604360056 604360056 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.0.10.10
t=0 0
m=audio 9702 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=ptime:20
a=sendrecv
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Here's the sip invite with the 'fix_nated_sdp' command:
--------------------------------------------------------------------------------------------------------------
INVITE sip:19167828326@xxx.xxx.xxx.xxx SIP/2.0
Record-Route: <sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes>
Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0
Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060
Max-Forwards: 69
From: "1009" <sip:1009@10.0.10.11>;tag=as49e00c81
To: <sip:19167828326@xxx.xxx.xxx.xxx>
Contact: <sip:1009@10.0.10.11:5060>
Call-ID: 4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060
<mailto:4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060>
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r356107
Date: Wed, 22 Feb 2012 03:18:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 347
P-hint: outbound
v=0
o=root 1009117068 1009117068 IN IP4 10.0.10.10
s=Asterisk PBX SVN-branch-1.8-r356107
c=IN IP4 10.50.50.8.10.0.10.10
t=0 0
m=audio 13540 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=ptime:20
a=sendrecv
a=oldmediaip:10.0.10.11
a=nortpproxy:yes
--------------------------------------------------------------------------------------------------------------
Is this a bug, or is it likely I have something else screwed up?
Thank you in advance for your assistance - this list is an incredible
resource!
-Ric
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla --http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/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
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
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3 comments
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participants (2)
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Daniel-Constantin Mierla -
Ric Marques