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: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
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
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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