[asterisk-bugs] [JIRA] (ASTERISK-29063) PJSIP leaves endpoint in Unavail state when successful qualification happens right after Unavail state change
Joshua C. Colp (JIRA)
noreply at issues.asterisk.org
Wed Sep 2 08:42:43 CDT 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-29063?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Joshua C. Colp updated ASTERISK-29063:
--------------------------------------
Assignee: Speed Dial Dave
Status: Waiting for Feedback (was: Triage)
This is due to the way that the PJSIP stack works. When we send a request we give it a timeout, if that timeout is met then we receive a callback saying no response. The SIP stack continues to attempt the transaction (it should probably stop, but that's within the stack itself) and subsequent responses are ignored (since we were told it already failed).
To try to overcome this is an improvement and would likely require modifications within the PJSIP stack itself with optional behavior changes. I don't think it would be easy to do.
As well, since this would be an improvement we generally don't accept those without a patch. Is it something you're looking into?
> PJSIP leaves endpoint in Unavail state when successful qualification happens right after Unavail state change
> -------------------------------------------------------------------------------------------------------------
>
> Key: ASTERISK-29063
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-29063
> Project: Asterisk
> Issue Type: Improvement
> Security Level: None
> Components: pjproject/pjsip
> Affects Versions: 16.11.0
> Environment: CentOS Linux 7.8.2003 x86_64
> Reporter: Speed Dial Dave
> Assignee: Speed Dial Dave
> Severity: Minor
>
> {noformat}[2020-08-28T14:07:11.295+0200] VERBOSE[14237] res_pjsip_logger.c: <--- Transmitting SIP request (417 bytes) to UDP:1.2.3.4:5060 --->
> OPTIONS sip:1.2.3.4 SIP/2.0
> Via: SIP/2.0/UDP 5.5.5.5:5060;rport;branch=z9hG4bKPj2cce2a99-8f1c-443c-8435-d5347147a76f
> From: <sip:trunkname at 5.5.5.5>;tag=b856990b-9ef9-45e7-92aa-573924e0bb4c
> To: <sip:1.2.3.4>
> Contact: <sip:trunkname at 5.5.5.5:5060>
> Call-ID: 254a56ce-1b12-4fd7-b5ff-2e7684341af4
> CSeq: 63071 OPTIONS
> Max-Forwards: 70
> User-Agent: Asterisk PBX 16.11.0
> Content-Length: 0
> [2020-08-28T14:07:11.795+0200] VERBOSE[2019] res_pjsip_logger.c: <--- Transmitting SIP request (417 bytes) to UDP:1.2.3.4:5060 --->
> OPTIONS sip:1.2.3.4 SIP/2.0
> Via: SIP/2.0/UDP 5.5.5.5:5060;rport;branch=z9hG4bKPj2cce2a99-8f1c-443c-8435-d5347147a76f
> From: <sip:trunkname at 5.5.5.5>;tag=b856990b-9ef9-45e7-92aa-573924e0bb4c
> To: <sip:1.2.3.4>
> Contact: <sip:trunkname at 5.5.5.5:5060>
> Call-ID: 254a56ce-1b12-4fd7-b5ff-2e7684341af4
> CSeq: 63071 OPTIONS
> Max-Forwards: 70
> User-Agent: Asterisk PBX 16.11.0
> Content-Length: 0
> [2020-08-28T14:07:12.795+0200] VERBOSE[2019] res_pjsip_logger.c: <--- Transmitting SIP request (417 bytes) to UDP:1.2.3.4:5060 --->
> OPTIONS sip:1.2.3.4 SIP/2.0
> Via: SIP/2.0/UDP 5.5.5.5:5060;rport;branch=z9hG4bKPj2cce2a99-8f1c-443c-8435-d5347147a76f
> From: <sip:trunkname at 5.5.5.5>;tag=b856990b-9ef9-45e7-92aa-573924e0bb4c
> To: <sip:1.2.3.4>
> Contact: <sip:trunkname at 5.5.5.5:5060>
> Call-ID: 254a56ce-1b12-4fd7-b5ff-2e7684341af4
> CSeq: 63071 OPTIONS
> Max-Forwards: 70
> User-Agent: Asterisk PBX 16.11.0
> Content-Length: 0
> [2020-08-28T14:07:14.295+0200] DEBUG[2019] res_pjsip.c: 0x7f0ea800c440: Internal tsx timer expired after 3000 msec
> [2020-08-28T14:07:14.295+0200] DEBUG[2019] res_pjsip.c: 0x7f0ea800c440: Timer handled here
> [2020-08-28T14:07:14.295+0200] DEBUG[2019] res_pjsip.c: 0x7f0ea800c440: Callbacks executed
> [2020-08-28T14:07:14.295+0200] VERBOSE[14237] res_pjsip/pjsip_configuration.c: Endpoint trunkname is now Unreachable
> [2020-08-28T14:07:14.295+0200] VERBOSE[14237] res_pjsip/pjsip_options.c: Contact trunkname/sip:1.2.3.4 is now Unreachable. RTT: 0.000 msec
> [2020-08-28T14:07:14.295+0200] DEBUG[1991] devicestate.c: No provider found, checking channel drivers for PJSIP - trunkname
> [2020-08-28T14:07:14.295+0200] DEBUG[14237] res_pjsip/pjsip_options.c: AOR 'trunkname' now has 0 available contacts
> [2020-08-28T14:07:14.295+0200] DEBUG[1991] devicestate.c: Changing state for PJSIP/trunkname - state 5 (Unavailable)
> [2020-08-28T14:07:14.295+0200] DEBUG[2050] app_queue.c: Device 'PJSIP/trunkname' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue.{noformat}
> The trunk is now marked Unavailable as it failed to respond within 3 seconds. Just half a second later a successful qualification attempt happens:
> {noformat}[2020-08-28T14:07:14.795+0200] VERBOSE[2019] res_pjsip_logger.c: <--- Transmitting SIP request (417 bytes) to UDP:1.2.3.4:5060 --->
> OPTIONS sip:1.2.3.4 SIP/2.0
> Via: SIP/2.0/UDP 5.5.5.5:5060;rport;branch=z9hG4bKPj2cce2a99-8f1c-443c-8435-d5347147a76f
> From: <sip:trunkname at 5.5.5.5>;tag=b856990b-9ef9-45e7-92aa-573924e0bb4c
> To: <sip:1.2.3.4>
> Contact: <sip:trunkname at 5.5.5.5:5060>
> Call-ID: 254a56ce-1b12-4fd7-b5ff-2e7684341af4
> CSeq: 63071 OPTIONS
> Max-Forwards: 70
> User-Agent: Asterisk PBX 16.11.0
> Content-Length: 0
> [2020-08-28T14:07:14.797+0200] VERBOSE[2019] res_pjsip_logger.c: <--- Received SIP response (351 bytes) from UDP:1.2.3.4:5060 --->
> SIP/2.0 480 No Routes Found
> Via: SIP/2.0/UDP 5.5.5.5:5060;received=5.5.5.5;branch=z9hG4bKPj2cce2a99-8f1c-443c-8435-d5347147a76f;rport=5060
> From: <sip:999999 at 5.5.5.5>;tag=b856990b-9ef9-45e7-92aa-573924e0bb4c
> To: <sip:1.2.3.4>;tag=aprqngfrt-f7hrai000gtv5
> Call-ID: 254a56ce-1b12-4fd7-b5ff-2e7684341af4
> CSeq: 63071 OPTIONS
> [2020-08-28T14:07:14.797+0200] DEBUG[2019] res_pjsip/pjsip_distributor.c: No dialog serializer for Response msg 480/OPTIONS/cseq=63071 (rdata0x7f0eec004368). Using request transaction as basis.
> [2020-08-28T14:07:14.797+0200] DEBUG[2019] res_pjsip/pjsip_distributor.c: Found transaction tsx0x7f0ea8018a38 for Response msg 480/OPTIONS/cseq=63071 (rdata0x7f0eec004368).
> [2020-08-28T14:07:14.797+0200] DEBUG[2019] res_pjsip/pjsip_distributor.c: Found serializer pjsip/options/trunkname-00000042 on transaction tsx0x7f0ea8018a38
> [2020-08-28T14:07:14.797+0200] DEBUG[14237] res_pjsip.c: 0x7f0ea800c440: PJSIP tsx response received
> [2020-08-28T14:07:14.797+0200] DEBUG[14237] res_pjsip.c: 0x7f0ea800c440: wrapper destroyed{noformat}
> At this point I would expect the trunk to immediately be brought back to Reachable/Avail state, but it stays Unavailable until the next qualification attempt ~27 seconds later:
> {noformat}[2020-08-28T14:07:41.295+0200] VERBOSE[30519] res_pjsip_logger.c: <--- Transmitting SIP request (417 bytes) to UDP:1.2.3.4:5060 --->
> OPTIONS sip:1.2.3.4 SIP/2.0
> Via: SIP/2.0/UDP 5.5.5.5:5060;rport;branch=z9hG4bKPjf7f88631-d669-44b4-8403-580006ad239d
> From: <sip:trunkname at 5.5.5.5>;tag=ab0b0636-9b7c-4a40-a38c-1c892ca68c08
> To: <sip:1.2.3.4>
> Contact: <sip:trunkname at 5.5.5.5:5060>
> Call-ID: c1d95174-50dd-47e2-bae4-426f8a1f4d63
> CSeq: 21729 OPTIONS
> Max-Forwards: 70
> User-Agent: Asterisk PBX 16.11.0
> Content-Length: 0
> [2020-08-28T14:07:41.297+0200] VERBOSE[2019] res_pjsip_logger.c: <--- Received SIP response (351 bytes) from UDP:1.2.3.4:5060 --->
> SIP/2.0 480 No Routes Found
> Via: SIP/2.0/UDP 5.5.5.5:5060;received=5.5.5.5;branch=z9hG4bKPjf7f88631-d669-44b4-8403-580006ad239d;rport=5060
> From: <sip:999999 at 5.5.5.5>;tag=ab0b0636-9b7c-4a40-a38c-1c892ca68c08
> To: <sip:1.2.3.4>;tag=aprqngfrt-sknglo2000l2e
> Call-ID: c1d95174-50dd-47e2-bae4-426f8a1f4d63
> CSeq: 21729 OPTIONS
> [2020-08-28T14:07:41.297+0200] DEBUG[2019] res_pjsip/pjsip_distributor.c: No dialog serializer for Response msg 480/OPTIONS/cseq=21729 (rdata0x7f0eec004368). Using request transaction as basis.
> [2020-08-28T14:07:41.297+0200] DEBUG[2019] res_pjsip/pjsip_distributor.c: Found transaction tsx0x7f0ea8018a38 for Response msg 480/OPTIONS/cseq=21729 (rdata0x7f0eec004368).
> [2020-08-28T14:07:41.297+0200] DEBUG[2019] res_pjsip/pjsip_distributor.c: Found serializer pjsip/options/trunkname-00000042 on transaction tsx0x7f0ea8018a38
> [2020-08-28T14:07:41.297+0200] DEBUG[14237] res_pjsip.c: 0x7f0ef00015b0: PJSIP tsx response received
> [2020-08-28T14:07:41.297+0200] DEBUG[14237] res_pjsip.c: 0x7f0ef00015b0: Cancelling timer
> [2020-08-28T14:07:41.297+0200] DEBUG[14237] res_pjsip.c: 0x7f0ef00015b0: Timer cancelled
> [2020-08-28T14:07:41.297+0200] DEBUG[14237] res_pjsip.c: 0x7f0ef00015b0: Callbacks executed
> [2020-08-28T14:07:41.297+0200] DEBUG[14237] res_pjsip.c: 0x7f0ef00015b0: wrapper destroyed
> [2020-08-28T14:07:41.297+0200] DEBUG[14237] res_pjsip/pjsip_options.c: Endpoint state compositor 'trunkname' is online as AOR 'trunkname' is available
> [2020-08-28T14:07:41.297+0200] VERBOSE[14237] res_pjsip/pjsip_configuration.c: Endpoint trunkname is now Reachable
> [2020-08-28T14:07:41.297+0200] VERBOSE[14237] res_pjsip/pjsip_options.c: Contact trunkname/sip:1.2.3.4 is now Reachable. RTT: 2.133 msec
> [2020-08-28T14:07:41.297+0200] DEBUG[14237] res_pjsip/pjsip_options.c: AOR 'trunkname' now has 1 available contacts
> [2020-08-28T14:07:41.297+0200] DEBUG[1991] devicestate.c: No provider found, checking channel drivers for PJSIP - trunkname
> [2020-08-28T14:07:41.297+0200] DEBUG[1991] devicestate.c: Changing state for PJSIP/trunkname - state 1 (Not in use)
> {noformat}
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