Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message
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:30003016094191500@10.4.4.251<sip%3A30003016094191500@10.4.4.251>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0 Max-Forwards: 70 Contact: <sip:22222238515000403@10.4.4.252<sip%3A22222238515000403@10.4.4.252>
From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7079@10.4.4.252 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:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7079@10.4.4.252 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:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060
;tag=cFS6jHj9DgjjF Call-ID: 1-7079@10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr: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:30003016094191500@10.4.4.251:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7079-1-0 From: 22222238515000403 <sip:22222238515000403@110.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060
;tag=cFS6jHj9DgjjF Call-ID: 1-7079@10.4.4.252 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:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060
;tag=cFS6jHj9DgjjF Call-ID: 1-7079@10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr: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:[service]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip];branch=[branch] Max-Forwards: 70 Contact: <sip:[field1]@[local_ip]> From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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 <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Date: Mon, 24 Aug 2009 15:42:31 -0500 Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:sipp@[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:
What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use. please send both. If we have to stop what we are doing to prove this works are you prepared to offer your soul to help document and other project maintenance? -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l 4000 scenario file: uac_redirect.xml FS dialplan: public.xml SIP trace: trace.log Here it is... sorry for not including at first... T. On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use. please send both.
If we have to stop what we are doing to prove this works are you prepared to offer your soul to help document and other project maintenance?
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@gmail.com> wrote:
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l 4000 scenario file: uac_redirect.xml FS dialplan: public.xml SIP trace: trace.log
The Via definition in your SIPp scenario differs between the INVITE and the ACK: INVITE: Via: SIP/2.0/[transport] [local_ip];branch=[branch] ACK: Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] In the INVITE, you are not adding the [local_port] as you do in the ACK. Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK. So it seems FS is not checking just the ACK's branch against the INVITE's; it seems it is checking the whole Via header. I don't know if this is in accordance to SIP specs. Another thing, about the way you are calling SIPp: do no use "-sn uac" and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx" means "use the internal (embedded) scenario named xxx". So this conflicts with the other parameter "-sf" which specifies an external profile. It seems this doesn't cause any problem (probably because in the sipp startup, -sf overrides -sn), but it is misleading. regards, takeshi
On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi@gmail.com> wrote:
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@gmail.com> wrote:
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l 4000 scenario file: uac_redirect.xml FS dialplan: public.xml SIP trace: trace.log
The Via definition in your SIPp scenario differs between the INVITE and the ACK:
INVITE: Via: SIP/2.0/[transport] [local_ip];branch=[branch]
ACK: Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
In the INVITE, you are not adding the [local_port] as you do in the ACK. Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK. So it seems FS is not checking just the ACK's branch against the INVITE's; it seems it is checking the whole Via header. I don't know if this is in accordance to SIP specs. Another thing, about the way you are calling SIPp: do no use "-sn uac" and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx" means "use the internal (embedded) scenario named xxx". So this conflicts with the other parameter "-sf" which specifies an external profile.
I mean, an external scenario (file). It seems this doesn't cause any problem (probably because in
the sipp startup, -sf overrides -sn), but it is misleading.
regards, takeshi
Hello Takeshi, Thanks for your hint... it worked out... so to be precise: VIA header of both INVITE and ACK messages MUST be identical (IP:PORT + branch)... and you are right... it might not be according to SIP specification. Anyhow, i get FS understand my ACK message. Finally, here is what i used and I'm getting some poor results .. but this is another topic :) Thanks for your help. Tihomir. sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000 <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="Basic Sipstone UAC"> <send retrans="500" start_rtd="1" start_rtd="2"> <![CDATA[ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 Contact: <sip:[field1]@[local_ip]> From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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" rtd="1"> </recv> <recv response="302" rtd="2"> </recv> <send> <![CDATA[ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi@gmail.com> wrote:
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@gmail.com> wrote:
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1
-l
4000 scenario file: uac_redirect.xml FS dialplan: public.xml SIP trace: trace.log
The Via definition in your SIPp scenario differs between the INVITE and the ACK:
INVITE: Via: SIP/2.0/[transport] [local_ip];branch=[branch]
ACK: Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
In the INVITE, you are not adding the [local_port] as you do in the ACK. Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK. So it seems FS is not checking just the ACK's branch against the INVITE's; it seems it is checking the whole Via header. I don't know if this is in accordance to SIP specs. Another thing, about the way you are calling SIPp: do no use "-sn uac" and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx" means "use the internal (embedded) scenario named xxx". So this conflicts with the other parameter "-sf" which specifies an external profile.
I mean, an external scenario (file).
It seems this doesn't cause any problem (probably because in
the sipp startup, -sf overrides -sn), but it is misleading.
regards, takeshi
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I beleive this is following the right rfc rules for dialog matching. If it is not, please open up a bug on jira.freeswitch.org with references of what exactly is not right. Mike On Aug 25, 2009, at 2:51 AM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
Hello Takeshi,
Thanks for your hint... it worked out... so to be precise:
VIA header of both INVITE and ACK messages MUST be identical (IP:PORT + branch)... and you are right... it might not be according to SIP specification. Anyhow, i get FS understand my ACK message.
Finally, here is what i used and I'm getting some poor results .. but this is another topic :)
Thanks for your help. Tihomir.
sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC"> <send retrans="500" start_rtd="1" start_rtd="2">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 Contact: <sip:[field1]@[local_ip]> From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag= [call_number] To: [service] <sip:[service]@[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" rtd="1"> </recv>
<recv response="302" rtd="2"> </recv>
<send> <![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch= [branch-3] From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag= [call_number] To: [service] <sip:[service]@[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>
On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi@gmail.com
wrote: On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi@gmail.com wrote: On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@gmail.com> wrote:
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf
test.txt -m 1 -l
4000 scenario file: uac_redirect.xml FS dialplan: public.xml SIP trace: trace.log
The Via definition in your SIPp scenario differs between the INVITE and the ACK:
INVITE: Via: SIP/2.0/[transport] [local_ip];branch=[branch]
ACK: Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
In the INVITE, you are not adding the [local_port] as you do in the ACK. Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK. So it seems FS is not checking just the ACK's branch against the INVITE's; it seems it is checking the whole Via header. I don't know if this is in accordance to SIP specs. Another thing, about the way you are calling SIPp: do no use "-sn uac" and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx" means "use the internal (embedded) scenario named xxx". So this conflicts with the other parameter "-sf" which specifies an external profile.
I mean, an external scenario (file).
It seems this doesn't cause any problem (probably because in
the sipp startup, -sf overrides -sn), but it is misleading.
regards, takeshi
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I wish I had a nickel for every guy struggling with sipp load testing vs real world traffic. On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
Hello Takeshi,
Thanks for your hint... it worked out... so to be precise:
VIA header of both INVITE and ACK messages MUST be identical (IP:PORT + branch)... and you are right... it might not be according to SIP specification. Anyhow, i get FS understand my ACK message.
Finally, here is what i used and I'm getting some poor results .. but this is another topic :)
Thanks for your help. Tihomir.
sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC"> <send retrans="500" start_rtd="1" start_rtd="2">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 Contact: <sip:[field1]@[local_ip]> From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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" rtd="1"> </recv>
<recv response="302" rtd="2"> </recv>
<send> <![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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>
On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi@gmail.com> wrote:
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@gmail.com> wrote:
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m
1 -l
4000 scenario file: uac_redirect.xml FS dialplan: public.xml SIP trace: trace.log
The Via definition in your SIPp scenario differs between the INVITE and the ACK:
INVITE: Via: SIP/2.0/[transport] [local_ip];branch=[branch]
ACK: Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
In the INVITE, you are not adding the [local_port] as you do in the ACK. Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK. So it seems FS is not checking just the ACK's branch against the INVITE's; it seems it is checking the whole Via header. I don't know if this is in accordance to SIP specs. Another thing, about the way you are calling SIPp: do no use "-sn uac" and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx" means "use the internal (embedded) scenario named xxx". So this conflicts with the other parameter "-sf" which specifies an external profile.
I mean, an external scenario (file).
It seems this doesn't cause any problem (probably because in
the sipp startup, -sf overrides -sn), but it is misleading.
regards, takeshi
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Well, you'd have another nickel from over here, then. If I can get this working before I'm tasked with something else I'll write up something more on the wiki about "Freeswitch and SIPp", but I'm not sure I'll get that chance. BB On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I wish I had a nickel for every guy struggling with sipp load testing vs real world traffic.
On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga@gmail.com>wrote:
Hello Takeshi,
Thanks for your hint... it worked out... so to be precise:
VIA header of both INVITE and ACK messages MUST be identical (IP:PORT + branch)... and you are right... it might not be according to SIP specification. Anyhow, i get FS understand my ACK message.
Finally, here is what i used and I'm getting some poor results .. but this is another topic :)
Thanks for your help. Tihomir.
sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC"> <send retrans="500" start_rtd="1" start_rtd="2">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 Contact: <sip:[field1]@[local_ip]> From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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" rtd="1"> </recv>
<recv response="302" rtd="2"> </recv>
<send> <![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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>
On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi@gmail.com> wrote:
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@gmail.com> wrote:
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m
1 -l
4000 scenario file: uac_redirect.xml FS dialplan: public.xml SIP trace: trace.log
The Via definition in your SIPp scenario differs between the INVITE and the ACK:
INVITE: Via: SIP/2.0/[transport] [local_ip];branch=[branch]
ACK: Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
In the INVITE, you are not adding the [local_port] as you do in the ACK. Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK. So it seems FS is not checking just the ACK's branch against the INVITE's; it seems it is checking the whole Via header. I don't know if this is in accordance to SIP specs. Another thing, about the way you are calling SIPp: do no use "-sn uac" and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx" means "use the internal (embedded) scenario named xxx". So this conflicts with the other parameter "-sf" which specifies an external profile.
I mean, an external scenario (file).
It seems this doesn't cause any problem (probably because in
the sipp startup, -sf overrides -sn), but it is misleading.
regards, takeshi
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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here's the one i use for making a call waiting x seconds and hanging up http://www.freeswitch.org/eg/load_test/dft_cap.xml This requires that the sipp terminate all the calls. careful with sipp, it's like a roach motel, you can get stuck trying to make it work and never get it to produce real-life situations. On Tue, Aug 25, 2009 at 3:23 PM, Bradley Brashier <bjbrashier@gmail.com>wrote:
Well, you'd have another nickel from over here, then. If I can get this working before I'm tasked with something else I'll write up something more on the wiki about "Freeswitch and SIPp", but I'm not sure I'll get that chance.
BB
On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I wish I had a nickel for every guy struggling with sipp load testing vs real world traffic.
On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga@gmail.com>wrote:
Hello Takeshi,
Thanks for your hint... it worked out... so to be precise:
VIA header of both INVITE and ACK messages MUST be identical (IP:PORT + branch)... and you are right... it might not be according to SIP specification. Anyhow, i get FS understand my ACK message.
Finally, here is what i used and I'm getting some poor results .. but this is another topic :)
Thanks for your help. Tihomir.
sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC"> <send retrans="500" start_rtd="1" start_rtd="2">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 Contact: <sip:[field1]@[local_ip]> From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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" rtd="1"> </recv>
<recv response="302" rtd="2"> </recv>
<send> <![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number] To: [service] <sip:[service]@[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>
On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi@gmail.com> wrote:
On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga@gmail.com> wrote:
sipp_cmd: sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt
-m 1 -l
4000 scenario file: uac_redirect.xml FS dialplan: public.xml SIP trace: trace.log
The Via definition in your SIPp scenario differs between the INVITE and the ACK:
INVITE: Via: SIP/2.0/[transport] [local_ip];branch=[branch]
ACK: Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
In the INVITE, you are not adding the [local_port] as you do in the ACK. Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK. So it seems FS is not checking just the ACK's branch against the INVITE's; it seems it is checking the whole Via header. I don't know if this is in accordance to SIP specs. Another thing, about the way you are calling SIPp: do no use "-sn uac" and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx" means "use the internal (embedded) scenario named xxx". So this conflicts with the other parameter "-sf" which specifies an external profile.
I mean, an external scenario (file).
It seems this doesn't cause any problem (probably because in
the sipp startup, -sf overrides -sn), but it is misleading.
regards, takeshi
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
... documentation hate that :)) ... but thats my life actually... thats what i do for living :) of course i can go forward and document a lot of things... a lot of docummentation is pending to completing the project i currently have using FS. well ... my soul? .. is it really necessary? T. On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use. please send both.
If we have to stop what we are doing to prove this works are you prepared to offer your soul to help document and other project maintenance?
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3] shouldn't that be Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
... documentation hate that :)) ... but thats my life actually... thats what i do for living :)
of course i can go forward and document a lot of things... a lot of docummentation is pending to completing the project i currently have using FS.
well ... my soul? .. is it really necessary?
T.
On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use. please send both.
If we have to stop what we are doing to prove this works are you prepared to offer your soul to help document and other project maintenance?
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0 http://sipp.sourceforge.net/doc3.0/reference.html *[branch]* - Provide a branch value which is a concatenation of magic cookie (z9hG4bK) + call number + message index in scenario. An offset (like [branch-N]) can be appended if you need to have the same branch value as a previous message. http://www.ietf.org/rfc/rfc3665.txt :according to examples in RFC i se branch being same for INVITE, 302 and ACK messages... [branch-3] makes branch of ACK message same as INVITE.... here is a test with just [branch] ... pls note: Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3 from ACK message .... the 3rd message... freeswitch@l01sipindir1> recv 530 bytes from udp/[10.4.4.252]:5060 at 23:11:34.506681: ------------------------------------------------------------------------ INVITE sip:30003016094191500@10.4.4.251<sip%3A30003016094191500@10.4.4.251>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0 Max-Forwards: 70 Contact: <sip:22222238515000403@10.4.4.252<sip%3A22222238515000403@10.4.4.252>
From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7093@10.4.4.252 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 23:11:34.506957: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7093@10.4.4.252 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Content-Length: 0 ------------------------------------------------------------------------ 2009-08-25 01:11:34.505272 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403@10.4.4.252:5060[7797f5ae-9103-11de-95e3-cda626a03c4b] 2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->30003016094191500 in context public 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## ServiceLookup ########################\n 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## contact = '' ##############\n 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## CallerNum = '38515000403' ##########\n 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth = '' ##########\n 2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->doRedirect in context public 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute redirect(sip:12345616094191500@pgw01.ot.hr:5060) send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:34.515143: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060
;tag=y38ac8m1m6e0K Call-ID: 1-7093@10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr: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 23:11:34.515369: ------------------------------------------------------------------------ ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3 From: 22222238515000403 <sip:22222238515000403@110.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060
;tag=y38ac8m1m6e0K Call-ID: 1-7093@10.4.4.252 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0
------------------------------------------------------------------------ 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute log(INFO ######################## RADIUS auth NOK!! ##########\n) 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth NOK!! ##########\n 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute hangup(USER_BUSY) 2009-08-25 01:11:34.513275 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/ 22222238515000403@10.4.4.252:5060 [CS_EXECUTE] [USER_BUSY] 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/22222238515000403@10.4.4.252:5060) Ended 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403@10.4.4.252:5060 [CS_DESTROY] send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:35.017059: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060
;tag=y38ac8m1m6e0K Call-ID: 1-7093@10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr: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
On Tue, Aug 25, 2009 at 12:55 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
shouldn't that be
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga@gmail.com>wrote:
... documentation hate that :)) ... but thats my life actually... thats what i do for living :)
of course i can go forward and document a lot of things... a lot of docummentation is pending to completing the project i currently have using FS.
well ... my soul? .. is it really necessary?
T.
On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use. please send both.
If we have to stop what we are doing to prove this works are you prepared to offer your soul to help document and other project maintenance?
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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Is there any way to make FS complain about what header is wrong ? T. On Tue, Aug 25, 2009 at 1:14 AM, Tihomir Culjaga <tculjaga@gmail.com> wrote:
this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0
http://sipp.sourceforge.net/doc3.0/reference.html
*[branch]* - Provide a branch value which is a concatenation of magic cookie (z9hG4bK) + call number + message index in scenario. An offset (like [branch-N]) can be appended if you need to have the same branch value as a previous message.
http://www.ietf.org/rfc/rfc3665.txt :according to examples in RFC i se branch being same for INVITE, 302 and ACK messages...
[branch-3] makes branch of ACK message same as INVITE....
here is a test with just [branch] ... pls note: Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3 from ACK message .... the 3rd message...
freeswitch@l01sipindir1> recv 530 bytes from udp/[10.4.4.252]:5060 at 23:11:34.506681: ------------------------------------------------------------------------ INVITE sip:30003016094191500@10.4.4.251<sip%3A30003016094191500@10.4.4.251>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0 Max-Forwards: 70 Contact: <sip:22222238515000403@10.4.4.252<sip%3A22222238515000403@10.4.4.252>
From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7093@10.4.4.252 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 23:11:34.506957: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060> Call-ID: 1-7093@10.4.4.252 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Content-Length: 0
------------------------------------------------------------------------ 2009-08-25 01:11:34.505272 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403@10.4.4.252:5060[7797f5ae-9103-11de-95e3-cda626a03c4b] 2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->30003016094191500 in context public 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## ServiceLookup ########################\n 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## contact = '' ##############\n 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## CallerNum = '38515000403' ##########\n 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth = '' ##########\n 2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->doRedirect in context public 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute redirect(sip:12345616094191500@pgw01.ot.hr:5060) send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:34.515143: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060
;tag=y38ac8m1m6e0K Call-ID: 1-7093@10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr: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 23:11:34.515369: ------------------------------------------------------------------------ ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3 From: 22222238515000403 <sip:22222238515000403@110.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060
;tag=y38ac8m1m6e0K Call-ID: 1-7093@10.4.4.252 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0
------------------------------------------------------------------------ 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute log(INFO ######################## RADIUS auth NOK!! ##########\n) 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ######################## RADIUS auth NOK!! ##########\n 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute hangup(USER_BUSY) 2009-08-25 01:11:34.513275 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/22222238515000403@10.4.4.252:5060 [CS_EXECUTE] [USER_BUSY] 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/22222238515000403@10.4.4.252:5060) Ended 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403@10.4.4.252:5060 [CS_DESTROY] send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:35.017059: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0 From: 22222238515000403 <sip:22222238515000403@10.4.4.252:5060>;tag=1 To: 30003016094191500 <sip:30003016094191500@10.4.4.251:5060
;tag=y38ac8m1m6e0K Call-ID: 1-7093@10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr: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
On Tue, Aug 25, 2009 at 12:55 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
shouldn't that be
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga@gmail.com>wrote:
... documentation hate that :)) ... but thats my life actually... thats what i do for living :)
of course i can go forward and document a lot of things... a lot of docummentation is pending to completing the project i currently have using FS.
well ... my soul? .. is it really necessary?
T.
On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
What is your exact sipp scenerio file and dialplan to this point after the changes you were suggested to use. please send both.
If we have to stop what we are doing to prove this works are you prepared to offer your soul to help document and other project maintenance?
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (5)
-
Anthony Minessale
-
Bradley Brashier
-
mayamatakeshi
-
Michael Jerris
-
Tihomir Culjaga