Re: [SR-Users] fix_nated_sdp issue

Daniel-Constantin Mierla Thu, 23 Feb 2012 23:15:18 -0800

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 callingfix_nated_sdp, and subsequent call:
# Routing to foreign domains

route[SIPOUT] {
xlog("---------------------- checking of outbound tosomewhere else -----------------------------------------");
                if (!uri==myself)

                {
xlog("<---------------------------------- Sending call out to someother 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 matchesport 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 otherdomain ------------------------------>
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 onentrance=(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/2.0100 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:19165551...@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 rtpproxyand nathelper, when the t_relay fires it errantly appends the addressfor 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 moduleupdating the sdp) with fix_nated_sdp() because of the way changes areapplied to the sip message. When using both, it results in concatenationof the two IP -- it is why I asked about the log, expecting you usedsuch two functions.

You may try using msg_apply_changes() in between such two functions, butI recommend making the config file decision in a way that you executeonly one such function. For example, you can set a flag after using suchfunctions and before using another one test that flag.


Cheers,
Daniel


Ric

*From:*Daniel-Constantin Mierla [mailto:mico...@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 (syslogmessages) 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 completelymisconfigured... I'm a total newb with Kamailio, working on a proof ofconcept 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 themedia ip address to resolve my audio issues, where 10.50.50.8 is theaddress outside my firewall. What I'm running into is the 'c=' linedoesn't get re-written properly... it inserts the specified address infront of the existing address, and I end up with the following line inmy INVITE:


c=IN IP4 10.50.50.810.0.10.10

I have the fix_nated_sdp command under route[sipout], because I onlywant to use it on calls being sent outside the nat firewall.


Here's the sip invite without the 'fix_nated_sdp' command:

--------------------------------------------------------------------------------------------------------------

INVITE sip:19165551...@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:19165551...@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:19167828...@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:19167828...@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 incredibleresource!


-Ric




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--
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@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@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] fix_nated_sdp issue

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