[SR-Users] No ACK from Cisco

Thu Jun 6 14:29:53 CEST 2013

Hi all!

I've got a problem with Kamailio<->Cisco<->PSTN.

Called from PSTN:

16:20:14.328786 IP (tos 0x80, ttl 255, id 0, offset 0, flags [none], proto UDP (17), length 1166)
    172.16.16.3.58446 > 172.16.17.8.sip: SIP, length: 1138
	INVITE sip:599674 at 172.16.17.8:5060 SIP/2.0
	Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9
	Remote-Party-ID: <sip:595311 at 172.16.16.3>;party=calling;screen=no;privacy=off
	From: <sip:595311 at 172.16.16.3>;tag=144D20C-8A7
	To: <sip:599674 at 172.16.17.8>
	Date: Thu, 06 Jun 2013 04:25:44 GMT
	Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4 at 172.16.16.3
	Supported: 100rel,timer,resource-priority,replaces
	Min-SE:  1800
	Cisco-Guid: 4251252826-3449229794-2149122083-881571867
	User-Agent: Cisco-SIPGateway/IOS-12.x
	Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
	CSeq: 101 INVITE
	Max-Forwards: 70
	Timestamp: 1370492744
	Contact: <sip:595311 at 172.16.16.3:5060>
	Expires: 180
	Allow-Events: telephone-event
	Content-Type: application/sdp
	Content-Disposition: session;handling=required
	Content-Length: 279

	v=0
	o=CiscoSystemsSIP-GW-UserAgent 6723 8551 IN IP4 172.16.16.3
	s=SIP Call
	c=IN IP4 172.16.16.3
	t=0 0
	m=audio 18550 RTP/AVP 8 18 101
	c=IN IP4 172.16.16.3
	a=rtpmap:8 PCMA/8000
	a=rtpmap:18 G729/8000
	a=fmtp:18 annexb=no
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16

16:20:14.329130 IP (tos 0x10, ttl 64, id 17361, offset 0, flags [none], proto UDP (17), length 327, bad cksum 0 (->bc99)!)
    172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 299
	SIP/2.0 100 Trying
	Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9
	From: <sip:595311 at 172.16.16.3>;tag=144D20C-8A7
	To: <sip:599674 at 172.16.17.8>
	Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4 at 172.16.16.3
	CSeq: 101 INVITE
	Server: kamailio (4.1.0-dev6 (x86_64/freebsd))
	Content-Length: 0

16:20:14.335619 IP (tos 0x10, ttl 64, id 17363, offset 0, flags [none], proto UDP (17), length 359, bad cksum 0 (->bc77)!)
    172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 331
	SIP/2.0 100 trying -- your call is important to us
	Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9
	From: <sip:595311 at 172.16.16.3>;tag=144D20C-8A7
	To: <sip:599674 at 172.16.17.8>
	Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4 at 172.16.16.3
	CSeq: 101 INVITE
	Server: kamailio (4.1.0-dev6 (x86_64/freebsd))
	Content-Length: 0

16:20:14.362576 IP (tos 0x10, ttl 64, id 17365, offset 0, flags [none], proto UDP (17), length 623, bad cksum 0 (->bb6d)!)
    172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 595
	SIP/2.0 180 Ringing
	Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9
	Record-Route: <sip:172.16.17.8;lr=on;did=558.b04>
	From: <sip:0074832595311 at 172.16.16.3>;tag=144D20C-8A7
	To: <sip:0074832599674 at 172.16.17.8>;tag=1054623052
	Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4 at 172.16.16.3
	CSeq: 101 INVITE
	Contact: <sip:0074832599674 at 10.120.0.18:32225;user=phone>
	Supported: replaces, path, timer, eventlist
	User-Agent: Grandstream GXV3140 1.0.7.76
	Allow-Events: talk, hold
	Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
	Content-Length: 0

16:20:23.621104 IP (tos 0x10, ttl 64, id 17380, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba50)!)
    172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865
	SIP/2.0 200 OK
	Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9
	Record-Route: <sip:172.16.17.8;lr=on;did=558.b04>
	From: <sip:0074832595311 at 172.16.16.3>;tag=144D20C-8A7
	To: <sip:0074832599674 at 172.16.17.8>;tag=1054623052
	Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4 at 172.16.16.3
	CSeq: 101 INVITE
	Contact: <sip:0074832599674 at 10.120.0.18:32225;user=phone>
	Supported: replaces, path, timer, eventlist
	User-Agent: Grandstream GXV3140 1.0.7.76
	Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
	Content-Type: application/sdp
	Content-Length:   266

	v=0
	o=0074832599674 8002 8000 IN IP4 10.120.0.18
	s=SIP Call
	c=IN IP4 10.120.0.18
	t=0 0
	m=audio 39206 RTP/AVP 8 18 101
	a=sendrecv
	a=rtpmap:8 PCMA/8000
	a=ptime:20
	a=rtpmap:18 G729/8000
	a=fmtp:18 annexb=no
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-15

16:20:24.126589 IP (tos 0x10, ttl 64, id 17383, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba4d)!)
    172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865
	SIP/2.0 200 OK

16:20:25.135006 IP (tos 0x10, ttl 64, id 17389, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba47)!)
    172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865
	SIP/2.0 200 OK

16:20:27.144412 IP (tos 0x10, ttl 64, id 17395, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba41)!)
    172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865
	SIP/2.0 200 OK

And no ACK from Cisco.

Is it Cisco config problem?

P.S. no sip-ua configuration, 

dial-peer voice 10002 voip
 description ** xxx **
 preference 1
 destination-pattern 5T
 voice-class codec 1
 session protocol sipv2
 session target ipv4:xxx
 session transport udp

I had no idea about no ACK. Maybe OK from Kamailio incorrect?

--
 WBR, Victor
  JID: coyote at bks.tv
  JID: coyote at bryansktel.ru
  I use FREE operation system: 3.9.4-calculate GNU/Linux