| Summary: | ASTERISK-21089: New SIP Channel Driver - Test Plan for Basic Calls | ||
| Reporter: | Matt Jordan (mjordan) | Labels: | Asterisk12 |
| Date Opened: | 2013-02-13 17:08:33.000-0600 | Date Closed: | 2013-05-10 08:13:02 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_pjsip |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Document on the wiki a set of scenarios that need to be tested for:
* Basic audio calls (inbound and outbound) * With/without authentication ** With the 'alwaysauthreject' kinds of logic * With/without endpoint identification ** With various forms of endpoint identification * PRACK/100 Rel * Session timers * Basic OPTIONS request handling * TCP/TLS transports * Ideally, realtime configuration (no sorcery object exists, but we should be thinking about this) This task isn't to write the tests, just get the plan up on the wiki. | ||
| Comments: | |||