It sounds like the vendor is handling NAT traversal on their side. They will be assuming that Asterisk is behind NAT, because of the presence of
private IP addresses – particularly in the contact, and will be rewriting various parts.
They may be able to disable this for you – otherwise you’ll need to rewrite the headers yourself.
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org]
On Behalf Of Nelson Migliaro
Sent: 17 October 2016 18:23
To: Kamailio (SER) - Users Mailing List <sr-users@lists.sip-router.org>
Subject: [SR-Users] BYE issue
Hello everybody,
I am having issues with one SIP vendor.
I have a Kamailio in bridge mode (private IP / Public IP) and some Asterisk and Media Gateways.
Calls get established and I have two way audio but when the remote party hangs up the call, the BYE arrives to the Kamailio and does not move forward.
I think the problem is SIP vendor rewrite the BYE header and change the asterisk IP with the public IP of the kamailio.
The IP that appears in the header of the BYE have to be the same that appears in the contact (UAC that send the call, in my case the Asterisk). Vendor should not change that IP. ¿Am I correct?
Thank you
-----------------------------------------------------------------------------------------------------
INVITE
----------------------------------------------------------------------------------------------------
2016/10/17 18:50:49.110967 PUBLIC-KAMAILIO-IP:5060 -> VENDOR-IP:6060
INVITE sip:DESTINATION-NUMBER@VENDOR-IP:6060 SIP/2.0
Record-Route: <sip:PUBLIC-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U
G9jYWw-;did=09b.9572;nat=yes>
Record-Route: <sip:PRIVATE-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U
G9jYWw-;did=09b.9572;nat=yes>
Via: SIP/2.0/UDP PUBLIC-KAMAILIO-IP;branch=z9hG4bK06a.07540d0e2f32a811ecf9c0a5235dc77a.1
Via: SIP/2.0/UDP PRIVATE-ASTERISK-IP:5060;received=PRIVATE-ASTERISK-IP;branch=z9hG4bK6bb5a7b3;rport=5060
Max-Forwards: 69
From: "SOURCE-NUMBER" <sip:SOURCE-NUMBER@MY-COMPANY>;tag=as5e87b96c
Contact: <sip:SOURCE-NUMBER@PRIVATE-ASTERISK-IP:5060>
Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060
CSeq: 102 INVITE
User-Agent: UAC
Date: Mon, 17 Oct 2016 16:53:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 292850421 292850421 IN IP4 PUBLIC-KAMAILIO-IP
s=Asterisk PBX
c=IN IP4 PUBLIC-KAMAILIO-IP
t=0 0
m=audio 23456 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
-----------------------------------------------------------------------------------------------------
BYE
-----------------------------------------------------------------------------------------------------
2016/10/17 18:50:58.241666 VENDOR-IP:6060 -> PUBLIC-KAMAILIO-IP:5060
BYE sip:SOURCE-NUMBER@PUBLIC-KAMAILIO-IP:5060 SIP/2.0
Via: SIP/2.0/UDP VENDOR-IP:6060;branch=z9hG4bKeff4.48943e76.0
Via: SIP/2.0/UDP VENDOR-IP:5060;branch=z9hG4bK1d4e605e4ll19f74fBYE421ce8658050206
Max-Forwards: 34
To: "SOURCE-NUMBER"<sip:SOURCE-NUMBER@YO>;tag=as5e87b96c
From: <sip:DESTINATION-NUMBER@PUBLIC-KAMAILIO-IP>;tag=421ce86-co1547-INS001
Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060
CSeq: 154701 BYE
User-Agent: VENDOR
Content-Length: 0
-----------------------------------------------------------------------------------------------------