[SR-Users] fix_nated_sdp issue

[Daniel-Constantin Mierla]

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 at 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 at 10.0.10.11>;tag=as5498b77e
>
> To: <sip:19165551212 at xxx.xxx.xxx.xxx>
>
> Contact: <sip:1009 at 10.0.10.11:5060>
>
> Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12 at 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 at 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 at 10.0.10.11>;tag=as49e00c81
>
> To: <sip:19167828326 at xxx.xxx.xxx.xxx>
>
> Contact: <sip:1009 at 10.0.10.11:5060>
>
> Call-ID: 4def5539675b6f644b99bb300e8ec8d6 at 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
>
>
>
> _______________________________________________
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-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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