Hello Brian, Dave
Still nothing... i've changed ip_addresses (remote_ip, local_ip) and changed branch within ACK message to meet INVITE's one....but it is still not enough... Also i checked RFC and this is how should it be ... (ACK without contact taking care to have correct TAGs and branch)... what can it be? ------------------------------------------------------------------------ INVITE sip:[email protected]<sip%[email protected]>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 Max-Forwards: 70 Contact: <sip:[email protected]<sip%[email protected]> > From: 22222238515000403 <sip:[email protected]:5060>;tag=1 To: 30003016094191500 <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 131 v=0 o=user1 53655765 2353687637 IN IP4 10.4.4.252 s=- c=IN IP4 10.4.4.252 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ------------------------------------------------------------------------ send 325 bytes to udp/[10.4.4.252]:5060 at 21:56:08.152812: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:[email protected]:5060>;tag=1 To: 30003016094191500 <sip:[email protected]:5060> Call-ID: [email protected] CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Content-Length: 0 ------------------------------------------------------------------------ send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.159929: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:[email protected]:5060>;tag=1 To: 30003016094191500 <sip:[email protected]:5060 >;tag=cFS6jHj9DgjjF Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:[email protected]:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 342 bytes from udp/[10.4.4.252]:5060 at 21:56:08.160166: ------------------------------------------------------------------------ ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:[email protected]:5060>;tag=1 To: 30003016094191500 <sip:[email protected]:5060 >;tag=cFS6jHj9DgjjF Call-ID: [email protected] CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.661299: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:[email protected]:5060>;tag=1 To: 30003016094191500 <sip:[email protected]:5060 >;tag=cFS6jHj9DgjjF Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:[email protected]:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 here is a scenario i use: <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="Basic Sipstone UAC"> <send retrans="500"> <![CDATA[ INVITE sip:[servi...@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip];branch=[branch] Max-Forwards: 70 Contact: <sip:[fiel...@[local_ip]> From: [field1] <sip:[fiel...@[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[servi...@[remote_ip]:[remote_port]> Call-ID: [call_id] CSeq: 1 INVITE Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" optional="true"> </recv> <recv response="302" rtd="true"> </recv> <send> <![CDATA[ ACK sip:[servi...@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] From: [field1] <sip:[fiel...@1[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[servi...@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 ]]> </send> <!-- definition of the response time repartition table (unit is ms) --> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> <!-- definition of the call length repartition table (unit is ms) --> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> </scenario> ---------- Forwarded message ---------- From: Brian West <[email protected]> To: [email protected] Date: Mon, 24 Aug 2009 15:42:31 -0500 Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message ACK sip:[servi...@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[servi...@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:s...@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 Use that.. your scenario has some hard coded IP's in the fields that shouldn't be there. /b On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote: Hello Brian, it doesn't work .. tried this today as well:
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