Re: [Sipp-users] crazy problem on simple call scenario
) || (strcmp(P_recv, NOTIFY) == 0) || (strcmp(P_recv, UPDATE) == 0) || (strcmp(P_recv, OPTIONS) == 0)) see on http://www.mail-archive.com/sipp-users@lists.sourceforge.net/msg03427.html Now, by using -aa , sipp answer properly to all OPTIONS NOTIFY and other kids with an 200 ok, but it answer to my all INVITE message too !! So this is not better for me... ritesh.gupta I tried to use in your way, but I haven't the same results !! Maybe it is because I have to enable OPTIONS - 200ok with -aa, could you please tell me what is your work around ? Thanks. Ruhi ASLAN Stagiaire ST40 - NOC/Operation VTX SERVICES SA Une société du groupe VTX Telecom Tél. direct : 021 721 12 18 Av. de Lavaux 101 - 1009 Pully http://www.vtx.ch http://www.vtx.ch/ - ruhi.as...@vtx-telecom.ch VTX, votre partenaire telecom proche de vous ! De : ritesh.gu...@bt.com [mailto:ritesh.gu...@bt.com] Envoyé : mercredi, 21. avril 2010 18:46 À : himanshu.ra...@gmail.com Cc : sipp-users@lists.sourceforge.net Objet : Re: [Sipp-users] crazy problem on simple call scenario I got the solution .. ... first run the UAC (registration) once registration are done then exit this SIPp instance. Then run new instance for UAS (Receiving Invite). Now it will work. From: Himanshu Rawat [mailto:himanshu.ra...@gmail.com] Sent: 20 April 2010 11:45 To: Gupta,R,Ritesh,DKH C Cc: s...@hirschbichler.biz; sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] crazy problem on simple call scenario Me too getting same error in the error log files even though I'm just running sipp as a client and connecting to the actual voice mail server. :( :(. Please anyone can tell why its happening.?? Cheers, Rawat On Tue, Apr 20, 2010 at 15:07, ritesh.gu...@bt.com wrote: Hi Michael, Can you please let me know how to split two UAC and UAS? Do we need to run two separate SIPp instance one for Register and one for Invite? In that case how they are going to map because Registration is done for particular number so how Invite instance going to understand that it should receive Invite for particular number? I tried two split UAC and UAS.. I run two separate sipp instance. Instance A for Register and Instance B for Invite. In that case also I am receiving same error on Instance A ---Discarding message which can't be mapped to a known.. Any suggestion any idea any help ? Thanks for support... Please find my XML for Instance A and Instance B Instance A XML scenario name=Basic Sipstone UAC !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword.-- send ![CDATA[ REGISTER sip:10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225 mailto:sip%3a...@10.230.53.225 From: 420sip:4...@10.230.53.225 mailto:sip%3a...@10.230.53.225 ;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Length: [len] ]] /send recv response=200 crlf=true /recv send ![CDATA[ SUBSCRIBE sip:4...@10.230.53.225 mailto:sip%3a...@10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225 mailto:sip%3a...@10.230.53.225 From: 420sip:4...@10.230.53.225 mailto:sip%3a...@10.230.53.225 ;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 4204o stamp 56125 Event: message-summary Content-Length: [len] ]] /send recv response=501 crlf=true /recv Instance B xml sample scenario name=Basic Sipstone UAS !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword.-- recv request=INVITE /recv send ![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: sip:[local_ip]:[local_port];transport=[transport] Content-Length: 0 ]] /send -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz] Sent: 12 April 2010 07:35 To: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] crazy problem on simple call scenario This scenario as described below won't work. If I understood the description correctly, the signalling-flow is UA Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- --INVITE- In sipp, the mapping
Re: [Sipp-users] crazy problem on simple call scenario
I got the solution .. ... first run the UAC (registration) once registration are done then exit this SIPp instance. Then run new instance for UAS (Receiving Invite). Now it will work. From: Himanshu Rawat [mailto:himanshu.ra...@gmail.com] Sent: 20 April 2010 11:45 To: Gupta,R,Ritesh,DKH C Cc: s...@hirschbichler.biz; sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] crazy problem on simple call scenario Me too getting same error in the error log files even though I'm just running sipp as a client and connecting to the actual voice mail server. :( :(. Please anyone can tell why its happening.?? Cheers, Rawat On Tue, Apr 20, 2010 at 15:07, ritesh.gu...@bt.commailto:ritesh.gu...@bt.com wrote: Hi Michael, Can you please let me know how to split two UAC and UAS? Do we need to run two separate SIPp instance one for Register and one for Invite? In that case how they are going to map because Registration is done for particular number so how Invite instance going to understand that it should receive Invite for particular number? I tried two split UAC and UAS.. I run two separate sipp instance. Instance A for Register and Instance B for Invite. In that case also I am receiving same error on Instance A ---Discarding message which can't be mapped to a known.. Any suggestion any idea any help ? Thanks for support... Please find my XML for Instance A and Instance B Instance A XML scenario name=Basic Sipstone UAC !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword.-- send ![CDATA[ REGISTER sip:10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060http://sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225mailto:sip%3a...@10.230.53.225 From: 420sip:4...@10.230.53.225mailto:sip%3a...@10.230.53.225;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Length: [len] ]] /send recv response=200 crlf=true /recv send ![CDATA[ SUBSCRIBE sip:4...@10.230.53.225mailto:sip%3a...@10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060http://sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225mailto:sip%3a...@10.230.53.225 From: 420sip:4...@10.230.53.225mailto:sip%3a...@10.230.53.225;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 4204o stamp 56125 Event: message-summary Content-Length: [len] ]] /send recv response=501 crlf=true /recv Instance B xml sample scenario name=Basic Sipstone UAS !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword.-- recv request=INVITE /recv send ![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: sip:[local_ip]:[local_port];transport=[transport] Content-Length: 0 ]] /send -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.bizmailto:s...@hirschbichler.biz] Sent: 12 April 2010 07:35 To: sipp-users@lists.sourceforge.netmailto:sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] crazy problem on simple call scenario This scenario as described below won't work. If I understood the description correctly, the signalling-flow is UA Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- --INVITE- In sipp, the mapping of a message (request/reply) is done by parsing for the SIP Call-ID - if a message is incoming with another call-id than the call-id in the originating request, the message is dropped as an unexpected message. In general, one sipp instance is not able to act as a UAC (for the registration process) and as an UAS (for the incomming invite request) at the same time. You have to split up the functionality to two sequenced sipp-instances: UA_C_ Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- and after that UA_S_ Proxy --INVITE- ---180 ---200 hth and br Michael On 2010-04-09 17:12, Ruhi Aslan wrote: *De :* Ruhi Aslan *Envoyé :* vendredi, 9. avril 2010 16:56 *À :* 'sipp-users-requ...@lists.sourceforge.netmailto:sipp-users-requ...@lists.sourceforge.net' *Objet :* help Hi all, Sipp is a great tool and I currently pull my hair out... I have some trouble with a very simple scenario. I even can't make a call to sipp registered phone. I first registered my phone : sipp -sf callee_hangup_process_test.xml
Re: [Sipp-users] crazy problem on simple call scenario
Hi Michael, Can you please let me know how to split two UAC and UAS? Do we need to run two separate SIPp instance one for Register and one for Invite? In that case how they are going to map because Registration is done for particular number so how Invite instance going to understand that it should receive Invite for particular number? I tried two split UAC and UAS.. I run two separate sipp instance. Instance A for Register and Instance B for Invite. In that case also I am receiving same error on Instance A ---Discarding message which can't be mapped to a known.. Any suggestion any idea any help ? Thanks for support... Please find my XML for Instance A and Instance B Instance A XML scenario name=Basic Sipstone UAC !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword.-- send ![CDATA[ REGISTER sip:10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225 From: 420sip:4...@10.230.53.225;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Length: [len] ]] /send recv response=200 crlf=true /recv send ![CDATA[ SUBSCRIBE sip:4...@10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225 From: 420sip:4...@10.230.53.225;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 4204o stamp 56125 Event: message-summary Content-Length: [len] ]] /send recv response=501 crlf=true /recv Instance B xml sample scenario name=Basic Sipstone UAS !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword.-- recv request=INVITE /recv send ![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: sip:[local_ip]:[local_port];transport=[transport] Content-Length: 0 ]] /send -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz] Sent: 12 April 2010 07:35 To: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] crazy problem on simple call scenario This scenario as described below won't work. If I understood the description correctly, the signalling-flow is UA Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- --INVITE- In sipp, the mapping of a message (request/reply) is done by parsing for the SIP Call-ID - if a message is incoming with another call-id than the call-id in the originating request, the message is dropped as an unexpected message. In general, one sipp instance is not able to act as a UAC (for the registration process) and as an UAS (for the incomming invite request) at the same time. You have to split up the functionality to two sequenced sipp-instances: UA_C_ Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- and after that UA_S_ Proxy --INVITE- ---180 ---200 hth and br Michael On 2010-04-09 17:12, Ruhi Aslan wrote: *De :* Ruhi Aslan *Envoyé :* vendredi, 9. avril 2010 16:56 *À :* 'sipp-users-requ...@lists.sourceforge.net' *Objet :* help Hi all, Sipp is a great tool and I currently pull my hair out... I have some trouble with a very simple scenario. I even can't make a call to sipp registered phone. I first registered my phone : sipp -sf callee_hangup_process_test.xml -inf csv/register_client.csv asterisk.ch -trace_err -r1 -m 1 ## register my sipp phone to get calls send ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputerip mailto:1...@mycomputerip CSeq: 1 REGISTER Contact: * Max-Forwards: 5 Expires: 0 User-Agent: SIPp/Linux Content-Length: 0 ]] /send recv response=404 optional=true next=1 /recv recv response=401 auth=true /recv *** Register Process *** send retrans=500 ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputerip mailto:1...@mycomputerip CSeq: 1 REGISTER Contact: * [AUTHENTICATION LINE] Max-Forwards: 5 Expires: 0 User-Agent: SIPp/Linux Content-Length: 0 ]] /send recv response=200 /recv ### phone
Re: [Sipp-users] crazy problem on simple call scenario
Me too getting same error in the error log files even though I'm just running sipp as a client and connecting to the actual voice mail server. :( :(. Please anyone can tell why its happening.?? Cheers, Rawat On Tue, Apr 20, 2010 at 15:07, ritesh.gu...@bt.com wrote: Hi Michael, Can you please let me know how to split two UAC and UAS? Do we need to run two separate SIPp instance one for Register and one for Invite? In that case how they are going to map because Registration is done for particular number so how Invite instance going to understand that it should receive Invite for particular number? I tried two split UAC and UAS.. I run two separate sipp instance. Instance A for Register and Instance B for Invite. In that case also I am receiving same error on Instance A ---Discarding message which can't be mapped to a known.. Any suggestion any idea any help ? Thanks for support... Please find my XML for Instance A and Instance B Instance A XML scenario name=Basic Sipstone UAC !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword. -- send ![CDATA[ REGISTER sip:10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225 sip%3a...@10.230.53.225 From: 420sip:4...@10.230.53.225 sip%3a...@10.230.53.225 ;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Length: [len] ]] /send recv response=200 crlf=true /recv send ![CDATA[ SUBSCRIBE sip:4...@10.230.53.225 sip%3a...@10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225 sip%3a...@10.230.53.225 From: 420sip:4...@10.230.53.225 sip%3a...@10.230.53.225 ;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 4204o stamp 56125 Event: message-summary Content-Length: [len] ]] /send recv response=501 crlf=true /recv Instance B xml sample scenario name=Basic Sipstone UAS !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword. -- recv request=INVITE /recv send ![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: sip:[local_ip]:[local_port];transport=[transport] Content-Length: 0 ]] /send -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz] Sent: 12 April 2010 07:35 To: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] crazy problem on simple call scenario This scenario as described below won't work. If I understood the description correctly, the signalling-flow is UA Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- --INVITE- In sipp, the mapping of a message (request/reply) is done by parsing for the SIP Call-ID - if a message is incoming with another call-id than the call-id in the originating request, the message is dropped as an unexpected message. In general, one sipp instance is not able to act as a UAC (for the registration process) and as an UAS (for the incomming invite request) at the same time. You have to split up the functionality to two sequenced sipp-instances: UA_C_ Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- and after that UA_S_ Proxy --INVITE- ---180 ---200 hth and br Michael On 2010-04-09 17:12, Ruhi Aslan wrote: *De :* Ruhi Aslan *Envoyé :* vendredi, 9. avril 2010 16:56 *À :* 'sipp-users-requ...@lists.sourceforge.net' *Objet :* help Hi all, Sipp is a great tool and I currently pull my hair out... I have some trouble with a very simple scenario. I even can't make a call to sipp registered phone. I first registered my phone : sipp -sf callee_hangup_process_test.xml -inf csv/register_client.csv asterisk.ch -trace_err -r1 -m 1 ## register my sipp phone to get calls send ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputerip mailto:1...@mycomputerip CSeq: 1 REGISTER Contact: * Max-Forwards: 5 Expires: 0 User-Agent: SIPp/Linux Content-Length: 0 ]] /send recv response=404 optional=true next=1 /recv recv response=401 auth=true /recv
Re: [Sipp-users] crazy problem on simple call scenario
The SIPp Client will be trying to match the CallID of calls in progress to any received packet. So either the CallID is not correct or the SIPp thinks the call has ended. Either a SIPp trace or an Ethereal trace should give you the answer. Peter Date: Tue, 20 Apr 2010 16:14:58 +0530 From: himanshu.ra...@gmail.com To: ritesh.gu...@bt.com CC: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] crazy problem on simple call scenario Me too getting same error in the error log files even though I'm just running sipp as a client and connecting to the actual voice mail server. :( :(. Please anyone can tell why its happening.?? Cheers, Rawat On Tue, Apr 20, 2010 at 15:07, ritesh.gu...@bt.com wrote: Hi Michael, Can you please let me know how to split two UAC and UAS? Do we need to run two separate SIPp instance one for Register and one for Invite? In that case how they are going to map because Registration is done for particular number so how Invite instance going to understand that it should receive Invite for particular number? I tried two split UAC and UAS.. I run two separate sipp instance. Instance A for Register and Instance B for Invite. In that case also I am receiving same error on Instance A ---Discarding message which can't be mapped to a known.. Any suggestion any idea any help ? Thanks for support... Please find my XML for Instance A and Instance B Instance A XML scenario name=Basic Sipstone UAC !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword.-- send ![CDATA[ REGISTER sip:10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225 From: 420sip:4...@10.230.53.225;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Length: [len] ]] /send recv response=200 crlf=true /recv send ![CDATA[ SUBSCRIBE sip:4...@10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: sip:4...@10.230.53.227:5060 To: 420sip:4...@10.230.53.225 From: 420sip:4...@10.230.53.225;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 4204o stamp 56125 Event: message-summary Content-Length: [len] ]] /send recv response=501 crlf=true /recv Instance B xml sample scenario name=Basic Sipstone UAS !-- In client mode (sipp placing calls), the Call-ID MUST be -- !-- generated by sipp. To do so, use [call_id] keyword.-- recv request=INVITE /recv send ![CDATA[ SIP/2.0 180 Ringing [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: sip:[local_ip]:[local_port];transport=[transport] Content-Length: 0 ]] /send -Original Message- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz] Sent: 12 April 2010 07:35 To: sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] crazy problem on simple call scenario This scenario as described below won't work. If I understood the description correctly, the signalling-flow is UA Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- --INVITE- In sipp, the mapping of a message (request/reply) is done by parsing for the SIP Call-ID - if a message is incoming with another call-id than the call-id in the originating request, the message is dropped as an unexpected message. In general, one sipp instance is not able to act as a UAC (for the registration process) and as an UAS (for the incomming invite request) at the same time. You have to split up the functionality to two sequenced sipp-instances: UA_C_ Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- and after that UA_S_ Proxy --INVITE- ---180 ---200 hth and br Michael On 2010-04-09 17:12, Ruhi Aslan wrote: *De :* Ruhi Aslan *Envoyé :* vendredi, 9. avril 2010 16:56 *À :* 'sipp-users-requ...@lists.sourceforge.net' *Objet :* help Hi all, Sipp is a great tool and I currently pull my hair out... I have some trouble with a very simple scenario. I even can't make a call to sipp registered phone. I first registered my phone : sipp -sf callee_hangup_process_test.xml -inf csv/register_client.csv asterisk.ch -trace_err -r1 -m 1 ## register my sipp phone to get calls send ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputerip mailto:1...@mycomputerip CSeq: 1
Re: [Sipp-users] crazy problem on simple call scenario
This scenario as described below won't work. If I understood the description correctly, the signalling-flow is UA Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- --INVITE- In sipp, the mapping of a message (request/reply) is done by parsing for the SIP Call-ID - if a message is incoming with another call-id than the call-id in the originating request, the message is dropped as an unexpected message. In general, one sipp instance is not able to act as a UAC (for the registration process) and as an UAS (for the incomming invite request) at the same time. You have to split up the functionality to two sequenced sipp-instances: UA_C_ Proxy ---REGISTER-- ---401--- ---REGISTER-- ---200--- and after that UA_S_ Proxy --INVITE- ---180 ---200 hth and br Michael On 2010-04-09 17:12, Ruhi Aslan wrote: *De :* Ruhi Aslan *Envoyé :* vendredi, 9. avril 2010 16:56 *À :* 'sipp-users-requ...@lists.sourceforge.net' *Objet :* help Hi all, Sipp is a great tool and I currently pull my hair out... I have some trouble with a very simple scenario. I even can't make a call to sipp registered phone. I first registered my phone : sipp -sf callee_hangup_process_test.xml -inf csv/register_client.csv asterisk.ch -trace_err -r1 -m 1 ## register my sipp phone to get calls send ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputerip mailto:1...@mycomputerip CSeq: 1 REGISTER Contact: * Max-Forwards: 5 Expires: 0 User-Agent: SIPp/Linux Content-Length: 0 ]] /send recv response=404 optional=true next=1 /recv recv response=401 auth=true /recv *** Register Process *** send retrans=500 ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputerip mailto:1...@mycomputerip CSeq: 1 REGISTER Contact: * [AUTHENTICATION LINE] Max-Forwards: 5 Expires: 0 User-Agent: SIPp/Linux Content-Length: 0 ]] /send recv response=200 /recv ### phone registered, sip show peer 44 tell me it's OK and reachable on mycomputerIP Then I ask to it to wait until an INVITE comes : recv request=INVITE crlf=true /recv In another window, I make a call with another phone number 43 ( correct scenarios and successfully tested ) sipp -sf callee_hangup.xml -inf csv/caller.cvs asterisk.ch -trace_err -r 1 -m 1 BUT, callee_hangup_process_test.xml doesn't get the INVITE from callee_hangup.xml scenario. The crazy thing is that wireshark says that it sends the expected INVITE to callee_hangup_process_test.xml ( on the right computer, on the right port ). But on my previous INVITE recv request, the count persist on 0 ! Here the INVITE sended to mycomputerIP ( supposed to make the INVITE recv reauest count up to 1 ) INVITE sip:4...@mycomputerip:5060 SIP/2.0 Record-Route: sip:sipproxy;lr=on;ftag=ftag;vsf=some...;did=... Via: SIP/2.0/UDP sipproxy;branch=z9hG4bK-ID2 Via: SIP/2.0/UDP asterisk.ch:5060;received=asterisk.ch;branch=z9hG4b-ID;rport=5060 From: 43 sip:4...@voip.vtx.ch;tag=as1cf8af76 To: sip:4...@mycomputerip:5060 Contact: sip:4...@_asterisk.ch_ Call-ID: call...@asterisk.ch mailto:call...@asterisk.ch CSeq: 102 INVITE User-Agent: voipua Max-Forwards: 69 Date: Fri, 09 Apr 2010 13:54:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 242 P-hint: outbound v=0 o=root 26199 26199 IN IP4 _asterisk.ch_ s=session c=IN IP4 _asterisk.ch_ t=0 0 m=audio 18150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - more info : I already use -aa option for OPTIONS NOTIFY request, and on the second OPTIONS, sipp crash on seg fault :-\ So where is my mistake ? Ruhi ASLAN Stagiaire ST40 - NOC/Operation -- Download Intel#174; Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] crazy problem on simple call scenario
Find example below. If that does not work then try to connect using any other sip phone such as xLite and take network traces for registration process then try to map the below message. It will definitely work. Some time you need to supply same [tag] as part of subscribe message which you got during 200ok response from server. --Registration (Send) -- 200 ok (Receive Response) --Subscribe (Send) send ![CDATA[ REGISTER sip:10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport Max-Forwards: 70 Contact: sip:te...@10.230.52.50:[local_port];rinstance=df769c75a8bad123 To: test2sip:te...@10.230.53.225 From: test2sip:te...@10.230.53.225;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Length: [len] ]] /send recv response=200 crlf=true /recv send ![CDATA[ SUBSCRIBE sip:te...@10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport Max-Forwards: 70 Contact: sip:te...@10.230.52.50:[local_port] To: test2sip:te...@10.230.53.225 From: test2sip:te...@10.230.53.225;tag=[call_number] Call-ID: [call_id] CSeq: [cseq] SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Event: message-summary Content-Length: [len] ]] /send recv response=501 crlf=true /recv From: Ruhi Aslan [mailto:ruhi.as...@vtx-telecom.ch] Sent: 09 April 2010 16:13 To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] crazy problem on simple call scenario De : Ruhi Aslan Envoyé : vendredi, 9. avril 2010 16:56 À : 'sipp-users-requ...@lists.sourceforge.net' Objet : help Hi all, Sipp is a great tool and I currently pull my hair out... I have some trouble with a very simple scenario. I even can't make a call to sipp registered phone. I first registered my phone : sipp -sf callee_hangup_process_test.xml -inf csv/register_client.csv asterisk.ch -trace_err -r1 -m 1 ## register my sipp phone to get calls send ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputeripmailto:1...@mycomputerip CSeq: 1 REGISTER Contact: * Max-Forwards: 5 Expires: 0 User-Agent: SIPp/Linux Content-Length: 0 ]] /send recv response=404 optional=true next=1 /recv recv response=401 auth=true /recv *** Register Process *** send retrans=500 ![CDATA[ REGISTER sip:sipproxy SIP/2.0 Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID From: sip:4...@mycomputerip;tag=1 To: sip:4...@mycomputerip Call-ID: 1...@mycomputeripmailto:1...@mycomputerip CSeq: 1 REGISTER Contact: * [AUTHENTICATION LINE] Max-Forwards: 5 Expires: 0 User-Agent: SIPp/Linux Content-Length: 0 ]] /send recv response=200 /recv ### phone registered, sip show peer 44 tell me it's OK and reachable on mycomputerIP Then I ask to it to wait until an INVITE comes : recv request=INVITE crlf=true /recv In another window, I make a call with another phone number 43 ( correct scenarios and successfully tested ) sipp -sf callee_hangup.xml -inf csv/caller.cvs asterisk.ch -trace_err -r 1 -m 1 BUT, callee_hangup_process_test.xml doesn't get the INVITE from callee_hangup.xml scenario. The crazy thing is that wireshark says that it sends the expected INVITE to callee_hangup_process_test.xml ( on the right computer, on the right port ). But on my previous INVITE recv request, the count persist on 0 ! Here the INVITE sended to mycomputerIP ( supposed to make the INVITE recv reauest count up to 1 ) INVITE sip:4...@mycomputerip:5060 SIP/2.0 Record-Route: sip:sipproxy;lr=on;ftag=ftag;vsf=some...;did=... Via: SIP/2.0/UDP sipproxy;branch=z9hG4bK-ID2 Via: SIP/2.0/UDP asterisk.ch:5060;received=asterisk.ch;branch=z9hG4b-ID;rport=5060 From: 43 sip:4...@voip.vtx.ch;tag=as1cf8af76 To: sip:4...@mycomputerip:5060 Contact: sip:4...@asterisk.ch Call-ID: call...@asterisk.chmailto:call...@asterisk.ch CSeq: 102 INVITE User-Agent: voipua Max-Forwards: 69 Date: Fri, 09 Apr 2010 13:54:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 242 P-hint: outbound v=0 o=root 26199 26199 IN IP4 asterisk.ch s=session c=IN IP4 asterisk.ch t=0 0 m=audio 18150 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - more info : I already use -aa option for OPTIONS NOTIFY request, and on the second OPTIONS, sipp crash on seg fault :-\ So where is my mistake ? Ruhi ASLAN Stagiaire ST40 - NOC/Operation VTX SERVICES SA Une société du groupe VTX Telecom Tél. direct : 021 721 12 18 Av. de Lavaux 101 - 1009 Pully http://www.vtx.chhttp://www.vtx.ch/ -
