Re: [Sipp-users] Why doesn´t SIPp re-send the ACK wh en it receive a lot of messages 200 OK?
This is not in 3.1, but is in the latest trunk versions. Charles Manish Sapariya <[EMAIL PROTECTED]> 10/10/2008 05:33 AM To ZiLi0n <[EMAIL PROTECTED]> cc sipp-users@lists.sourceforge.net Subject Re: [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a lot of messages 200 OK? Hi, Out of curiosity, I tried this (-trace_calldebug) flag and sipp complaint that no such flag available. # sipp -v SIPp v3.1-TLS-PCAP, version svn9910, built May 2 2008, 15:31:41. Am I using the latest version? Thanks and Regards, Manish ZiLi0n wrote: > Mr. Charles, I have seen "hash 0" in calldebug when SIPp sends the message "BYE" : > > 2008-10-1008:12:14:7851223619134.785339 Set Retransmission Hash: 0 (recv index 3, send index 9) > 2008-10-1008:12:14:7851223619134.785348 Processing message 10 of type 3 for call [EMAIL PROTECTED] at 183076. > 2008-10-1008:12:15:2951223619135.295335 Processing message 10 of type 3 for call [EMAIL PROTECTED] at 183586. > 2008-10-1008:12:15:2951223619135.295344 Retransmisison required (1 retransmissions, max 9) > 2008-10-1008:12:15:2951223619135.295362 Sending UDP message for call [EMAIL PROTECTED] (index 9, hash 502009980): > > Thanks a lot! > >> To: [EMAIL PROTECTED] >> From: [EMAIL PROTECTED] >> Date: Thu, 9 Oct 2008 11:42:54 -0400 >> CC: sipp-users@lists.sourceforge.net >> Subject: Re: [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a lot of messages 200 OK? >> >> Ths is a -trace_msg trace not a -trace_calldebug trace. >> >> Charles >> >> >> >> >> ZiLi0n <[EMAIL PROTECTED]> >> 10/09/2008 11:27 AM >> >> To >> >> cc >> >> Subject >> [Sipp-users] Re: Why doesn´t SIPp re-send the ACK when it receive a lot >> of messages 200 OK? >> >> >> >> >> >> >> Mr Charles. Thank. Here is the traces >> >> In this traces, SIPp don´t resends the ACK even when Asterisk resends the >> message '200 OK': >> >> --- 2008-10-08 >> 13:11:08:995.378 >> UDP message received [758] bytes : >> >> SIP/2.0 200 OK^M >> Via: SIP/2.0/UDP >> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M >> From: sipp ;tag=2882SIPpTag00230^M >> To: sut ;tag=as7fc476b0^M >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 INVITE^M >> User-Agent: Asterisk PBX 1.6.0^M >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M >> Supported: replaces, timer^M >> Contact: ^M >> Content-Type: application/sdp^M >> Content-Length: 261^M >> ^M >> v=0^M >> o=root 377264768 377264768 IN IP4 192.168.1.246^M >> s=Asterisk PBX 1.6.0^M >> c=IN IP4 192.168.1.246^M >> t=0 0^M >> m=audio 18590 RTP/AVP 8 101^M >> a=rtpmap:8 PCMA/8000^M >> a=rtpmap:101 telephone-event/8000^M >> a=fmtp:101 0-16^M >> a=silenceSupp:off - - - -^M >> a=ptime:20^M >> a=sendrecv^M >> >> --- 2008-10-08 >> 13:11:08:995.480 >> UDP message sent (383 bytes): >> >> ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0^M >> Via: SIP/2.0/UDP 192.168.1.179:5060;branch=z9hG4bK-2882-230-4^M >> From: sipp ;tag=2882SIPpTag00230^M >> To: sut ;tag=as7fc476b0^M >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 ACK^M >> Contact: sip:[EMAIL PROTECTED]:5060^M >> Max-Forwards: 70^M >> Subject: Performance Test^M >> Content-Length: 0^M >> ^M >> --- 2008-10-08 >> 13:11:09:993.684 >> UDP message received [758] bytes : >> >> SIP/2.0 200 OK^M >> Via: SIP/2.0/UDP >> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M >> From: sipp ;tag=2882SIPpTag00230^M >> To: sut ;tag=as7fc476b0^M >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 INVITE^M >> User-Agent: Asterisk PBX 1.6.0^M >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M >> Supported: replaces, timer^M >> Contact: ^M >> Content-Type: application/sdp^M >> Content-Length: 261^M >> ^M >> v=0^M >> o=root 377264768 377264768 IN IP4 192.168.1.246^M >> s=Asterisk PBX 1.6.0^M >> c=IN IP4 192.168.1.246^M >> t=0 0^M >> m=audio 18590 RTP/AVP 8 101^M >> a=rtpmap:8 PCMA/8000^M >> a=rtpmap:101 telephone-event/8000^M >> a=fmtp:101 0-16^M >> a=silenceSupp:off - - - -^M >> a=ptime:20^M >> a=sendrecv^M >> >> --- 2008-10-08 >> 13:11:10:994.064 >> UDP message received [758] bytes : >> >> SIP/2.0 200 OK^M >> Via: SIP/2.0/UDP >> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M >> From: sipp ;tag=2882SIPpTag00230^M >> To: sut ;tag=as7fc476b0^M >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 INVITE^M >> User-Agent: Asterisk PBX 1.6.0^M >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M >> Supported: replaces, timer^M >> Contact: ^M >> Content-Type: application/sdp^M >> Content-Length: 261^M >> ^M >> v=0^M >> o=root 377264768 377264768 IN IP4 192.168.1.246^M >> s=Asterisk PBX 1.6.0^M >> c=IN IP4 192.168.1.246^M >> t=0 0^M >> m=audio 18590 RTP/AVP 8 101^M >> a=rtpmap:8 PCMA/8000^M >> a=rtpmap:101 telephone-event/8000^M >> >> >> Other problem i
Re: [Sipp-users] Why doesn´t SIPp re-send the ACK wh en it receive a lot of messages 200 OK?
Hi, Out of curiosity, I tried this (-trace_calldebug) flag and sipp complaint that no such flag available. # sipp -v SIPp v3.1-TLS-PCAP, version svn9910, built May 2 2008, 15:31:41. Am I using the latest version? Thanks and Regards, Manish ZiLi0n wrote: > Mr. Charles, I have seen "hash 0" in calldebug when SIPp sends the message > "BYE" : > > 2008-10-1008:12:14:7851223619134.785339 Set Retransmission Hash: 0 > (recv index 3, send index 9) > 2008-10-1008:12:14:7851223619134.785348 Processing message 10 of type > 3 for call [EMAIL PROTECTED] at 183076. > 2008-10-1008:12:15:2951223619135.295335 Processing message 10 of type > 3 for call [EMAIL PROTECTED] at 183586. > 2008-10-1008:12:15:2951223619135.295344 Retransmisison required (1 > retransmissions, max 9) > 2008-10-1008:12:15:2951223619135.295362 Sending UDP message for call > [EMAIL PROTECTED] (index 9, hash 502009980): > > Thanks a lot! > >> To: [EMAIL PROTECTED] >> From: [EMAIL PROTECTED] >> Date: Thu, 9 Oct 2008 11:42:54 -0400 >> CC: sipp-users@lists.sourceforge.net >> Subject: Re: [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a >> lot of messages 200 OK? >> >> Ths is a -trace_msg trace not a -trace_calldebug trace. >> >> Charles >> >> >> >> >> ZiLi0n <[EMAIL PROTECTED]> >> 10/09/2008 11:27 AM >> >> To >> >> cc >> >> Subject >> [Sipp-users] Re: Why doesn´t SIPp re-send the ACK when it receive a lot >> of messages 200 OK? >> >> >> >> >> >> >> Mr Charles. Thank. Here is the traces >> >> In this traces, SIPp don´t resends the ACK even when Asterisk resends the >> message '200 OK': >> >> --- 2008-10-08 >> 13:11:08:995.378 >> UDP message received [758] bytes : >> >> SIP/2.0 200 OK^M >> Via: SIP/2.0/UDP >> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M >> From: sipp ;tag=2882SIPpTag00230^M >> To: sut ;tag=as7fc476b0^M >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 INVITE^M >> User-Agent: Asterisk PBX 1.6.0^M >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M >> Supported: replaces, timer^M >> Contact: ^M >> Content-Type: application/sdp^M >> Content-Length: 261^M >> ^M >> v=0^M >> o=root 377264768 377264768 IN IP4 192.168.1.246^M >> s=Asterisk PBX 1.6.0^M >> c=IN IP4 192.168.1.246^M >> t=0 0^M >> m=audio 18590 RTP/AVP 8 101^M >> a=rtpmap:8 PCMA/8000^M >> a=rtpmap:101 telephone-event/8000^M >> a=fmtp:101 0-16^M >> a=silenceSupp:off - - - -^M >> a=ptime:20^M >> a=sendrecv^M >> >> --- 2008-10-08 >> 13:11:08:995.480 >> UDP message sent (383 bytes): >> >> ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0^M >> Via: SIP/2.0/UDP 192.168.1.179:5060;branch=z9hG4bK-2882-230-4^M >> From: sipp ;tag=2882SIPpTag00230^M >> To: sut ;tag=as7fc476b0^M >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 ACK^M >> Contact: sip:[EMAIL PROTECTED]:5060^M >> Max-Forwards: 70^M >> Subject: Performance Test^M >> Content-Length: 0^M >> ^M >> --- 2008-10-08 >> 13:11:09:993.684 >> UDP message received [758] bytes : >> >> SIP/2.0 200 OK^M >> Via: SIP/2.0/UDP >> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M >> From: sipp ;tag=2882SIPpTag00230^M >> To: sut ;tag=as7fc476b0^M >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 INVITE^M >> User-Agent: Asterisk PBX 1.6.0^M >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M >> Supported: replaces, timer^M >> Contact: ^M >> Content-Type: application/sdp^M >> Content-Length: 261^M >> ^M >> v=0^M >> o=root 377264768 377264768 IN IP4 192.168.1.246^M >> s=Asterisk PBX 1.6.0^M >> c=IN IP4 192.168.1.246^M >> t=0 0^M >> m=audio 18590 RTP/AVP 8 101^M >> a=rtpmap:8 PCMA/8000^M >> a=rtpmap:101 telephone-event/8000^M >> a=fmtp:101 0-16^M >> a=silenceSupp:off - - - -^M >> a=ptime:20^M >> a=sendrecv^M >> >> --- 2008-10-08 >> 13:11:10:994.064 >> UDP message received [758] bytes : >> >> SIP/2.0 200 OK^M >> Via: SIP/2.0/UDP >> 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M >> From: sipp ;tag=2882SIPpTag00230^M >> To: sut ;tag=as7fc476b0^M >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 INVITE^M >> User-Agent: Asterisk PBX 1.6.0^M >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M >> Supported: replaces, timer^M >> Contact: ^M >> Content-Type: application/sdp^M >> Content-Length: 261^M >> ^M >> v=0^M >> o=root 377264768 377264768 IN IP4 192.168.1.246^M >> s=Asterisk PBX 1.6.0^M >> c=IN IP4 192.168.1.246^M >> t=0 0^M >> m=audio 18590 RTP/AVP 8 101^M >> a=rtpmap:8 PCMA/8000^M >> a=rtpmap:101 telephone-event/8000^M >> >> >> Other problem is SIPp retrans the message BYE even when Asterisk has sent >> the '200 OK': >> >> U +0.026130 192.168.1.136:5060 -> 192.168.1.246:5060 >> BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-12878-227-9 >> From: sipp ;tag=12878SIPpTag00227 >> To:
Re: [Sipp-users] Why doesn´t SIPp re-send the ACK wh en it receive a lot of messages 200 OK?
Ths is a -trace_msg trace not a -trace_calldebug trace. Charles ZiLi0n <[EMAIL PROTECTED]> 10/09/2008 11:27 AM To cc Subject [Sipp-users] Re: Why doesn´t SIPp re-send the ACK when it receive a lot of messages 200 OK? Mr Charles. Thank. Here is the traces In this traces, SIPp don´t resends the ACK even when Asterisk resends the message "200 OK": --- 2008-10-08 13:11:08:995.378 UDP message received [758] bytes : SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M From: sipp ;tag=2882SIPpTag00230^M To: sut ;tag=as7fc476b0^M Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE^M User-Agent: Asterisk PBX 1.6.0^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces, timer^M Contact: ^M Content-Type: application/sdp^M Content-Length: 261^M ^M v=0^M o=root 377264768 377264768 IN IP4 192.168.1.246^M s=Asterisk PBX 1.6.0^M c=IN IP4 192.168.1.246^M t=0 0^M m=audio 18590 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off - - - -^M a=ptime:20^M a=sendrecv^M --- 2008-10-08 13:11:08:995.480 UDP message sent (383 bytes): ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.179:5060;branch=z9hG4bK-2882-230-4^M From: sipp ;tag=2882SIPpTag00230^M To: sut ;tag=as7fc476b0^M Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK^M Contact: sip:[EMAIL PROTECTED]:5060^M Max-Forwards: 70^M Subject: Performance Test^M Content-Length: 0^M ^M --- 2008-10-08 13:11:09:993.684 UDP message received [758] bytes : SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M From: sipp ;tag=2882SIPpTag00230^M To: sut ;tag=as7fc476b0^M Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE^M User-Agent: Asterisk PBX 1.6.0^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces, timer^M Contact: ^M Content-Type: application/sdp^M Content-Length: 261^M ^M v=0^M o=root 377264768 377264768 IN IP4 192.168.1.246^M s=Asterisk PBX 1.6.0^M c=IN IP4 192.168.1.246^M t=0 0^M m=audio 18590 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off - - - -^M a=ptime:20^M a=sendrecv^M --- 2008-10-08 13:11:10:994.064 UDP message received [758] bytes : SIP/2.0 200 OK^M Via: SIP/2.0/UDP 192.168.1.179:5060;branch=z9hG4bK-2882-230-0;received=192.168.1.179^M From: sipp ;tag=2882SIPpTag00230^M To: sut ;tag=as7fc476b0^M Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE^M User-Agent: Asterisk PBX 1.6.0^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces, timer^M Contact: ^M Content-Type: application/sdp^M Content-Length: 261^M ^M v=0^M o=root 377264768 377264768 IN IP4 192.168.1.246^M s=Asterisk PBX 1.6.0^M c=IN IP4 192.168.1.246^M t=0 0^M m=audio 18590 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M Other problem is SIPp retrans the message BYE even when Asterisk has sent the "200 OK": U +0.026130 192.168.1.136:5060 -> 192.168.1.246:5060 BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-12878-227-9 From: sipp ;tag=12878SIPpTag00227 To: sut ;tag=as7e62485f Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 # U +0.000146 192.168.1.246:5060 -> 192.168.1.136:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-12878-227-9;received=192.168.1.136 From: sipp ;tag=12878SIPpTag00227 To: sut ;tag=as7e62485f Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 U +0.152849 192.168.1.136:5060 -> 192.168.1.246:5060 BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-12878-227-9 From: sipp ;tag=12878SIPpTag00227 To: sut ;tag=as7e62485f Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 # U +0.000125 192.168.1.246:5060 -> 192.168.1.136:5060 SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-12878-227-9;received=192.168.1.136 From: sipp ;tag=12878SIPpTag00227 To: sut ;tag=as7e62485f Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX 1.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 I think that SIPp CLIENT hasn´t got time for receive the response from Asterisk. this errors only occurrs when the load is heavy. Some solution? PD: For this test I use sipp.2008-09-25.tar Thanks. > To: [EMAIL PROTECTED] > From: [EMAIL PROTECTED] > Dat
Re: [Sipp-users] Why doesn´t SIPp re-send the ACK wh en it receive a lot of messages 200 OK?
You should include the output of -trace_calldebug in your message. Charles ZiLi0n <[EMAIL PROTECTED]> wrote on 10/07/2008 04:33:10 PM: > > > SIPp should retransmit the ACK if the retransmitted 200 is identical. You > > should enable -trace_msg and -trace_calldebug and see what the logs tell > > you. > > The retrasnmitted meesages 200 OK are identicals. > > I´m using SIPp with Asterisk.Asterisk show this errors: > Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Maximum > retries exceeded on transmission 778f89593967725f0abe40eb1752504c (at) > 10.10.206.53 for seqno 1620 (Critical Response) > > Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Hanging > up call 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 no reply to > our critical packet. > > A solution? Thanks > > > > > Charles > > > > > > > > > > ZiLi0n <[EMAIL PROTECTED]> > > 10/05/2008 02:26 PM > > > > To > > > > cc > > > > Subject > > [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a lot of > > messages 200 OK? > > > > > > > > > > > > > > I am testing my Asterisk Server. This is my configuration: > > > > SIPpCLIENT --- Asterisk Server --- SIPpSERVER > > > > If the simultaneous calls is less than 300 the test is OK, but when the > > simultaneous calls is approximately 350 calls, the calls hang up: > > > > Asterisk sends to SIPpCLIENT the message "200 OK". SIPpCLIENT sends to > > Asterisk the message "ACK". > > Asterisk re-sends to SIPpCLIENT the message "200 OK". > > SIPpCLIENT doesn´t re-send the message "ACK" to Asterisk. > > > > I think that message "ACK" is lost... but the cpu load Asterisk Server is > > approximately 50% and the net status is perfect. > > > > Why does not SIPpCLIENT re-send the ACK to Asterisk? Asterisk needs the > > ACK for complete the call > > > > Thank´s > > > > La cartera, las gafas. ¿te falta algo? Ahora llévate Messenger en tu móvil > > - > > 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=100&url=/ > > ___ > > 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'schallenge > > 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=100&url=/ > > ___ > > Sipp-users mailing list > > Sipp-users@lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/sipp-users > > Ahora llévate lo mejor de MSN y Windows Live, en tu móvil > - > 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=100&url=/ > ___ > 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=100&url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Why doesn´t SIPp re-send the ACK wh en it receive a lot of messages 200 OK?
SIPp should retransmit the ACK if the retransmitted 200 is identical. You should enable -trace_msg and -trace_calldebug and see what the logs tell you. Charles ZiLi0n <[EMAIL PROTECTED]> 10/05/2008 02:26 PM To cc Subject [Sipp-users] Why doesn´t SIPp re-send the ACK when it receive a lot of messages 200 OK? I am testing my Asterisk Server. This is my configuration: SIPpCLIENT --- Asterisk Server --- SIPpSERVER If the simultaneous calls is less than 300 the test is OK, but when the simultaneous calls is approximately 350 calls, the calls hang up: Asterisk sends to SIPpCLIENT the message "200 OK". SIPpCLIENT sends to Asterisk the message "ACK". Asterisk re-sends to SIPpCLIENT the message "200 OK". SIPpCLIENT doesn´t re-send the message "ACK" to Asterisk. I think that message "ACK" is lost... but the cpu load Asterisk Server is approximately 50% and the net status is perfect. Why does not SIPpCLIENT re-send the ACK to Asterisk? Asterisk needs the ACK for complete the call Thank´s La cartera, las gafas. ¿te falta algo? Ahora llévate Messenger en tu móvil - 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=100&url=/ ___ 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=100&url=/ ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
