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 To: sip:DESTINATION-NUMBER@VENDOR-IP 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 Route: sip:PUBLIC-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1 Route: sip:PRIVATE-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1 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
-----------------------------------------------------------------------------------------------------
Nelson,
You are very correct. The request URI of the BYE (and all other in-dialog requests, such as reinvites) should equal the Contact URI of the party to which it is being sent, and this should not change even if the in-dialog request is being sent through Kamailio because of Record-Route.
-- Alex
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 To: sip:DESTINATION-NUMBER@VENDOR-IP 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 Route: sip:PUBLIC-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1 Route: sip:PRIVATE-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1 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
-----------------------------------------------------------------------------------------------------
Handling NAT, perhaps, but not correctly. The RURI doesn't change. But not does it have to determine where the request is actually sent on the network and transport layers.
-- Alex
-- Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
Is it not just the case that the vendor does not support loose routing?
-----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Alex Balashov Sent: 18 October 2016 08:40 To: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Subject: Re: [SR-Users] BYE issue
Handling NAT, perhaps, but not correctly. The RURI doesn't change. But not does it have to determine where the request is actually sent on the network and transport layers.
-- Alex
-- Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus.
_______________________________________________ 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
Thank you very much for your support,
When you say: "They may be able to disable this for you – otherwise you’ll need to rewrite the headers yourself."
How can I rewrite the header if I dont have destination IP?, there are four Asterisk servers and all of them send calls to the bridged Kamailio and I dont have Asterisk private IP in the BYE request.
Regards and again thank you,
Nelson.-
2016-10-18 9:18 GMT+02:00 Phil Lavin phil.lavin@cloudcall.com:
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=AAAAAAQEAw8MDgsAAHYAcVddXkZWRV VDVl1MMDIudm9pY2U
G9jYWw-;did=09b.9572;nat=yes>
Record-Route: <sip:PRIVATE-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf= AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRV VDVl1MMDIudm9pY2U
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
To: sip:DESTINATION-NUMBER@VENDOR-IP
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=z9hG4bK1d4e605e4ll19f74fBYE421 ce8658050206
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
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Phil Lavin