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Summary:ASTERISK-12215: Asterisk Fails SIP Performance Test with Navtel Interwatch 9500 Hardware Call Generator
Reporter:Alisher (licedey)Labels:
Date Opened:2008-06-17 13:40:23Date Closed:2008-06-17 13:41:06
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
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Description:- We are facing a lot of issues when we try to test Asterisk on HW call generator Navtel Interwatch. Navtel Interwatch is an enterprise grade call generator, Telco usually use it test softswitch and ip pbx equipment.

1) During the dialing, Asterisk sends Notify message on Ring, Interwatch cannot understand by throwing error.

2) Call test fails with 5060 port, other ports works fine. We have tried different IP PBX, only Asterisk fails the test with 5060 port. The problem is that Asterisk send local ip address of caller instead of ip pbx domain.
Need to patch chan_sip.c to fix it.

3) Local Extensions Test: 3CPS, 100 peers, 27 concurrent calls 72 hours
  Result: 40% Skipped calls 60% Successful calls

4) SIP Trunk: Two Asterisk Boxes with each 100 peers and 1 SIP Trunks
 Interwatch --> Asterisk PBX ----> SW ---> Asterisk Box---> Interwatch

Error messages:

18 00:34:58] WARNING[28035] chan_sip.c: Maximum retries exceeded on transmission AA4OQn__CEYAABez08hfSQ--c36495@xener.com for seqno 34384980 (Critical Response)
[Jun 18 00:36:15] WARNING[28035] chan_sip.c: Hanging up call AA1rC3__CJMAABez08hfSQ--c36621@xener.com - no reply to our critical packet.
RTCP SR transmission error  rtcp halted Operation not permitted


****** ADDITIONAL INFORMATION ******

Tested with Asterisk versions: 1.2, 1.4.19, and latest 1.4.21.
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