| Summary: | ASTERISK-28982: res_pjsip_t38: Does not resume as audio when negotiation fails | ||||
| Reporter: | Christian Berger (Christian_Berger) | Labels: | |||
| Date Opened: | 2020-07-07 08:06:23 | Date Closed: | |||
| Priority: | Minor | Regression? | No | ||
| Status: | Open/New | Components: | Resources/res_fax Resources/res_pjsip_t38 | ||
| Versions: | 13.32.0 16.11.1 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Attachments: | ( 0) bugreport.pcap ( 1) extensions.conf ( 2) messages ( 3) pjsip.conf | |||
| Description: | We operate Asterisk as a B2Bua in our network. It's task is to transcode, write CDRs and act as a T38 gateway. We are currently trying to move over to PJSIP from the legacy SIP stack.
We have noticed the following potential bug: When T38 is negotiated on a call leg, voice traffic to it will stop. However even if T38 is rejected it will not send anything via RTP. This issue sounds like ASTERISK-28441, however the patch there does not appear to change the problem. In fact the changed function does not seem to be called at all. We have, however, found one potential bug in t38_fallback_response_cb. Provisional Response codes 1xx are handled as failures. However this doesn't seem to solve the problem. | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2020-07-07 08:06:24.732-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. By: Joshua C. Colp (jcolp) 2020-07-07 08:11:14.760-0500 Can you elaborate why you think this should be treated as a separate issue despite ASTERISK-28441 existing? It's the same failure result. Are you just concerned that if the other one is fixed it may not cover this specific case? By: Christian Berger (Christian_Berger) 2020-07-07 08:12:30.522-0500 If you filter the pcap for ip.dst==213.167.160.122, you will see that the voice stream stops after packet 249. This is just as the T.38 invite is being sent in packet 252. Even when this INVITE is rejected in packets 257 and 258 the RTP-stream does not resume. By: Christian Berger (Christian_Berger) 2020-07-07 08:16:10.098-0500 @Joshua: This issue certainly looks is simmilar. The workaround patch provided with the other issue does not appear to change the situation at all, while in the other issue it seemed to solve the problem reliably. By: Joshua C. Colp (jcolp) 2020-07-07 08:18:56.669-0500 Right, the problem is likely something further down that exhibits itself under other scenarios with T.38 negotiation and the workaround is just that - a workaround for a particular scenario. | ||||