Re: [Sipp-users] Caller scenario sends out REGISTER packets without respecting -users or -l flag
There is no way to limit transactions or requests; only calls (either with -l or -users). If your call has only one concurrent transaction (probably the only way for SIPp to work correctly); then the number of calls is an upper bound on transactions. You can disable retransmissions with -nr to prevent more than one request in the same transaction; but that is not going to give you an accurate workload. If a call fails (i.e. the INVITE is never replied to); then that call is replaced with a new one that sends register. You can limit the total number of calls with -m 100. Charles Manish Sapariya [EMAIL PROTECTED] wrote on 09/22/2008 06:16:19 AM: Hi All, I am trying to create a work load where in I want to have 100 max established calls after the system has reached count of 100 calls. My caller scenario is approximately as follows: === Send Register Expect proxy auth Send Register with auth Expect 200 OK Send Invite Expect Proxy auth Send Invite with auth expect OK play pcap file wait for the duration of pcap file Send Bye Expect OK = If my server under test sends the response to both register and Invite within time for all 100 requests, everything works just fine. However, if for some reason, my server fails to send reply to some of the invite packets, then sipp keeps on sending register packets irrespective of how many total register packet it has sent. In this way it keeps bombarding my server with register packets, and server fails to send the reply to the invite packet. I am sure there is a problem with server, however question to the list is that, Is it possible to tell sipp that keep at the max 100 outstanding register request or invite request. I tried using -l and -users option. However both of this do not take un-acknowledged register and invite request into account. Please let me know if I need to provide more info or clarification. I can share the scenario and the exact command line it that helps. Thanks and Regards, Manish - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] SIP-I message format in sipp
This is applied as revision 536. It would be great if we had a nice string structure throughout the code so that we could handle non-null terminated strings both on send and receive for all types of messages; but this is a good start. Charles Andy Aicken [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 09/09/2008 07:04 PM To 'darshan b n' [EMAIL PROTECTED], sipp-users@lists.sourceforge.net cc Subject Re: [Sipp-users] SIP-I message format in sipp Hi Darshan, I created a patch for handling SIP-I messages, as the current message handling in SIPp treats everything as a string so doesn?t handle ISUP message bodies that contain a binary \x00. This ends up being treated as a string termination resulting in message gets truncated. The patch is available at: https://sourceforge.net/tracker/?func=detailatid=637566aid=1965508group_id=104305 It needs more rigorous testing but worked ok for me with the type of functionality I was using. Regards Andy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of darshan b n Sent: 04 September 2008 12:30 To: sipp-users@lists.sourceforge.net Subject: [Sipp-users] SIP-I message format in sipp Hi all , i want know how to create a SIP-I message in sipp please respond with a sample message format Thanks darshan On 04/09/2008, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Send Sipp-users mailing list submissions to sipp-users@lists.sourceforge.net To subscribe or unsubscribe via the World Wide Web, visit https://lists.sourceforge.net/lists/listinfo/sipp-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Sipp-users digest... Today's Topics: 1. Re: Force source IP source Port at IP layer (Klaus Darilion) 2. sipp remote (RTP) port handling (Jan Rudinsk?) -- Message: 1 Date: Thu, 04 Sep 2008 10:05:02 +0200 From: Klaus Darilion [EMAIL PROTECTED] Subject: Re: [Sipp-users] Force source IP source Port at IP layer To: Cyrille OLIVIER [EMAIL PROTECTED] Cc: sipp-users@lists.sourceforge.net Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed FYI: If you want to change the src IP you can also use this patch: https://sourceforge.net/tracker/?func=detailatid=637566aid=1823593group_id=104305 klaus Cyrille OLIVIER schrieb: Dear sipp-users, Again, I asked my requests about SIPp client using TCP: Is it possible to force sipp to use specific IP source Port source, at IP layer, for send messages when TCP with single socket (option '-t t1' used) ? I tried many things: -bind_local: seems unuseful. -i x.x.x.x -p options: it's only for some SIP headers but not for IP packet header. send -source_ip=x.x.x.x -source_port= for INVITE message look for this subject in mailing list archives ... Currently, I don't know which other workaround or things to do :( I would really appreciate any help about that Thanks a lot, BR, Cyrille Discutez gratuitement avec vos amis en vid?o ! T?l?chargez Messenger, c'est gratuit ! http://www.windowslive.fr/majmessenger.asp - Sponsored by: SourceForge.net Community Choice Awards: VOTE NOW! Studies have shown that voting for your favorite open source project, along with a healthy diet, reduces your potential for chronic lameness and boredom. Vote Now at http://www.sourceforge.net/community/cca08 ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Message: 2 Date: Thu, 04 Sep 2008 12:32:12 +0200 From: Jan Rudinsk? [EMAIL PROTECTED] Subject: [Sipp-users] sipp remote (RTP) port handling To: sipp-users@lists.sourceforge.net Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-2 Hi, I'm using SIPp to generate a call with RTP media. Media are sent to remote side, recorded and sent back. However SIPp sends media to a different remote port than offered by the remote side. SIPp: SIP INVITE with SDP m=audio 6000 RTP/AVP 0 Remote: 200 OK with SDP m=audio 18436 RTP/AVP 0 101 SIP:RTP incoming on 6000(OK) Remote: RTP incoming on 1843(instead of 18436) Attached: scenario graph, packet capture Does anyone know the solution? Thank you, JaR -- Ing. Jan Rudinsky Czech Technical University in Prague Cesnet z.s.p.o. RD Centre (RDC) for Mobile Applications [EMAIL PROTECTED]
Re: [Sipp-users] Caller scenario sends out REGISTER packets without respecting -users or -l flag
Hi Charles, Thanks for the clarification. I have some (may be very basic) queries: - What is the basic difference between transaction/request AND calls? - Why sipp is not considering sending of INVITE as a call establishment process? - How does people typically deal with such issue OR is it that I have unique situation here? Thanks for all the help. -Manish Charles P Wright wrote: There is no way to limit transactions or requests; only calls (either with -l or -users). If your call has only one concurrent transaction (probably the only way for SIPp to work correctly); then the number of calls is an upper bound on transactions. You can disable retransmissions with -nr to prevent more than one request in the same transaction; but that is not going to give you an accurate workload. If a call fails (i.e. the INVITE is never replied to); then that call is replaced with a new one that sends register. You can limit the total number of calls with -m 100. Charles Manish Sapariya [EMAIL PROTECTED] wrote on 09/22/2008 06:16:19 AM: Hi All, I am trying to create a work load where in I want to have 100 max established calls after the system has reached count of 100 calls. My caller scenario is approximately as follows: === Send Register Expect proxy auth Send Register with auth Expect 200 OK Send Invite Expect Proxy auth Send Invite with auth expect OK play pcap file wait for the duration of pcap file Send Bye Expect OK = If my server under test sends the response to both register and Invite within time for all 100 requests, everything works just fine. However, if for some reason, my server fails to send reply to some of the invite packets, then sipp keeps on sending register packets irrespective of how many total register packet it has sent. In this way it keeps bombarding my server with register packets, and server fails to send the reply to the invite packet. I am sure there is a problem with server, however question to the list is that, Is it possible to tell sipp that keep at the max 100 outstanding register request or invite request. I tried using -l and -users option. However both of this do not take un-acknowledged register and invite request into account. Please let me know if I need to provide more info or clarification. I can share the scenario and the exact command line it that helps. Thanks and Regards, Manish - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] -rsa issue fixes in sipp v3.1?
hey all, I have see a couple post about -rsa option not working in sipp v3.1. I have the same problem as well and wondering when the issue will be fixed. to repeat the problem: sipp is running as uas. when phone receives a sip request from a device under test (DUT), by default it sends the reply to the receiving port (usually a random UDP port), not to the port in request's contact header. in older releases (v2.0), -rsa overrides the default ip/port. However this is not working in v3.1 thus all my testing failed. Does anyone have a fix already? thanks. == Make it as simple as possible, but not simpler. -Albert Einstein - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Possible bug with SIPp pertaining to uac rtp ports
Hello all, I'm encountering a problem when trying to run tests with Asterisk. I have created an Asterisk config such that: SIPp (uac_pcap) Asterisk - SIPp (uas) SIP traffic is flowing perfectly, however, I can't get RTP media to flow. The media is leaving the uac_pcap instance of SIPp and then is rejected by the Asterisk server with an ICMP packet saying that the destination UDP port is not open. The RTP port, to the best of my knowledge, is negotiated (and generally dictated by the server) at the time of call construction via SIP. At this point, I figured either that Asterisk was sending an incorrect UDP port for media, or SIPp was ignoring this port. After some tcpdumps and asterisk sip debugs, it looks to be the latter. Asterisk is sending the SIPp client a five digit UDP port number, and SIPp is then sending media to a four digit port, which is essentially the five digit port with the last digit (1s digit) truncated. Here is the output I am seeing: Asterisk output specifying the RTP port: Audio is at 192.168.100.25 port 14272 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP pbx*CLI --- Reliably Transmitting (NAT) to 192.168.100.156:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.156:5060;branch=z9hG4bK-16464-1-0;received= 192.168.100.156 From: sipp sip:[EMAIL PROTECTED]:5060;tag=16464SIPpTag091 To: sut sip:[EMAIL PROTECTED]:5060;tag=as462c2628 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 242 v=0 o=root 2124 2124 IN IP4 192.168.100.25 s=session c=IN IP4 192.168.100.25 t=0 0 m=audio 14272 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv RTP traffic leaving SIPp computer(192.168.100.156) and going to asterisk( 192.168.100.25): 14:11:03.125895 IP 192.168.100.156.6000 192.168.100.25.1427: UDP, length 252 14:11:03.156076 IP 192.168.100.156.6000 192.168.100.25.1427: UDP, length 252 14:11:03.186269 IP 192.168.100.156.6000 192.168.100.25.1427: UDP, length 252 So 14272 was specified as the port, but SIPp is sending it to 14272. Here are the two listings of parameters that I am using to start SIPp: UAC: sudo ./sipp -s s -sn uac_pcap -p 5060 -i 192.168.100.156 192.168.100.25 -r 0 -l 1 -mi 192.168.100.156 UAS: ./sipp -sn uas -p 5061 -mp 6001 -mi 192.168.100.156 -i 192.168.100.156-rtp_echo Is this a bug, or am I doing something wrong? Ideas? - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
[Sipp-users] Variable interpolation in regexp
Hello, is it possible to interpolate variables, fields and keywords to a regexp? I would like to do something like this: ereg regexp=sip:[EMAIL PROTECTED]:[local_port];expires=([0-9]+) search_in=hdr header=Contact: assign_to=dummy,expires check_it=true / But it doesn't work: Failed regexp match: looking in ' sip:[EMAIL PROTECTED]:6060 ;expires=40;received=sip:192.168.88.29:6060', with regexp 'sip:[EMAIL PROTECTED]:[local_port];expires=([0-9]+)'. - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Caller scenario sends out REGISTER packets without respecting -users or -l flag
Hi Charles, -nr did the trick for time being. Thanks for the help. -Manish Charles P Wright wrote: There is no way to limit transactions or requests; only calls (either with -l or -users). If your call has only one concurrent transaction (probably the only way for SIPp to work correctly); then the number of calls is an upper bound on transactions. You can disable retransmissions with -nr to prevent more than one request in the same transaction; but that is not going to give you an accurate workload. If a call fails (i.e. the INVITE is never replied to); then that call is replaced with a new one that sends register. You can limit the total number of calls with -m 100. Charles Manish Sapariya [EMAIL PROTECTED] wrote on 09/22/2008 06:16:19 AM: Hi All, I am trying to create a work load where in I want to have 100 max established calls after the system has reached count of 100 calls. My caller scenario is approximately as follows: === Send Register Expect proxy auth Send Register with auth Expect 200 OK Send Invite Expect Proxy auth Send Invite with auth expect OK play pcap file wait for the duration of pcap file Send Bye Expect OK = If my server under test sends the response to both register and Invite within time for all 100 requests, everything works just fine. However, if for some reason, my server fails to send reply to some of the invite packets, then sipp keeps on sending register packets irrespective of how many total register packet it has sent. In this way it keeps bombarding my server with register packets, and server fails to send the reply to the invite packet. I am sure there is a problem with server, however question to the list is that, Is it possible to tell sipp that keep at the max 100 outstanding register request or invite request. I tried using -l and -users option. However both of this do not take un-acknowledged register and invite request into account. Please let me know if I need to provide more info or clarification. I can share the scenario and the exact command line it that helps. Thanks and Regards, Manish - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users - This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
