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

 



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