Yasin,
1. The Request URI of the ACK must match the Contact of the other end (the Contact URI in the 200 OK), was you correctly surmised. This includes all attributes of the Contact URI, including port.
2. The ACK you pasted needs to contain Route: headers whose values match the Record-Route headers in the 200 OK. This is not the case in the ACK you pasted.
3. The construction of the ACK is entirely up to the calling User Agent; if it is malformed, this is not something Kamailio can fix.
-- Alex
It would be relevant to see the 200ok as received by each hop in the call path. Also, be sure you don't use fix_nated_contact() on the proxy if it is not the first node next to endpoint -- anyhow it is recommended to use set_contact_alias().
As a clafication, do you use tcp/tls between Kamailio2 and Asterisk?
Cheers, Daniel
On 12/04/16 22:12, Yasin CANER wrote:
Hello; before sending this email i searched on google and doesnt solve this issue. all call flows are correct but one call that this isnt working right. it sends to _wrong port_ to ACK for 200 OK. i tried to fix contact header or remove contact header but it wasnt work. i looked at ietf for ACK and couldnt figure out why it happens. Does it need add a record route or remove Contact Header for every ack ?
Thanks for help.
i figure out Kamailio-2 adds a Route header to ACK packet for sending Kamailio-1:5060, even if it doesnt add any command for it in cfg.
Here is call flow
Asterisk and Kamailio is on the same ip and machine and public ip. different are ports. Kamailio-1 is another machine
INVITE : UAC1-----> Kamailio-1:5060----->Kamailio-2:5061--->Asterisk:5060
200 OK: UAC1<----- Kamailio-1:5060<-----Kamailio-2:5061<---Asterisk:5060
ACK : UAC1-----> Kamailio-1:5060----->Kamailio-2:5061--->Asterisk:10432
200 OK: UAC1<----- Kamailio-1:5060<-----Kamailio-2:5061<---Asterisk:5060
ACK : UAC1-----> Kamailio-1:5060----->Kamailio-2:5061--->Asterisk:10432
Retransmission.......
Here is ACK packet, is it about_port on RU?_
Asteriskip:5060----------->Kamailio2-ip:5061
SIP/2.0 200 OK Via: SIP/2.0/UDP kamailio2-ip:5061;branch=z9hG4bK4d2f.837f45483db59574c4dd70f70f8d0099.0;received=kamailio2-ip;rport=5061 Via: SIP/2.0/UDP kamailio1-ip-main;branch=z9hG4bK4d2f.fb781e8caa5f96426c82530a23c0cc97.0 Via: SIP/2.0/UDP uacip:5060;received=uacip;branch=z9hG4bK143c7384;rport=10432 Record-Route: sip:kamailio2-ip:5061;lr;ftag=as1c529e28;did=304.35c1;vsf=AAAAAAoBAQMLBQ4GAQN2A3kDFgQYABYeGRoyMDM- Record-Route: sip:kamailio1-ip-main;lr;ftag=as1c529e28 From: 903122977162 sip:903122977162@kamailio2-ip;tag=as1c529e28 To: sip:03129110911@tstxyz.netgsm.com.tr;tag=as39358508 Call-ID: 169f342556f4e956445dfe0e2ee8ecf2@uacip:5060 CSeq: 103 INVITE Server: sipgw2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:10213129110911@kamailio2-ip:5060 Content-Type: application/sdp Require: timer Content-Length: 305
v=0 o=root 455426546 455426546 IN IP4 kamailio2-ip s=Asterisk PBX 11.21.2 c=IN IP4 kamailio2-ip t=0 0 m=audio 15926 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Kamailio2-ip:5061----> Asterisk:10432 ACK sip:10213129110911@kamailio2-ip:10432 SIP/2.0 Via: SIP/2.0/UDP kamailio2-ip:5061;branch=z9hG4bK4d2f.deb7bd96617c7067212dbc1673a216d2.0 Via: SIP/2.0/UDP kamailio1-ip-main;branch=z9hG4bK4d2f.acd6ee83df5babb9e8f53268a4b1b948.0 Via: SIP/2.0/UDP uacip:5060;received=uacip;branch=z9hG4bK59e2e846;rport=10432 Max-Forwards: 68 From: sip:903122977162@kamailio2-ip;tag=as1c529e28 To: sip:03129110911@tstxyz.netgsm.com.tr;tag=as39358508 Call-ID: 169f342556f4e956445dfe0e2ee8ecf2@uacip:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 11.21.2 Content-Length: 0
ietf :
If the INVITE request whose response is being acknowledged had Route header fields, those header fields MUST appear in the ACK. This is to ensure that the ACK can be routed properly through any downstream stateless proxies.
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
Hello; Thanks for advices. Kamailio2 and Asterisk use UDP. Well, i fixed issue that Client Firewall has SIP alg and changes the ContactUri port and disabled SIP ALG.
First,i try to add Record Route to ACK on kamailio2 But it didnt work and Transaction couldnt complete for BYE Second, tried to fix Contact or removed on Kamailio2 , it didnt work.
Third , tried to fix R-URI port changes to 5060 that asterisk's listen port. it worked.
After comparing Elastix packets and Kamailio-1 packets , i realized that Firewall changes SIP packets with SIP ALG.
in conclusion , i need to work NATDETECT route on Kamailio 1 and Kamailio 2 that i can fix this issue on Kamailio.
Thanks.
Call Flow
Contact Header Elastix:192.168.0.0:5060--->FW-->Elastix:PublicIP:10432-->Kamailio1-->Kamailio2+Asterisk.
-- View this message in context: http://sip-router.1086192.n5.nabble.com/Misrouting-ACK-packet-for-200-Ok-mes... Sent from the Users mailing list archive at Nabble.com.
-
Alex Balashov -
Daniel-Constantin Mierla -
Yasin CANER -
ycaner