Summary: | ASTERISK-28070: testsuite: Sniffer assumes pjmedia will use ports below 10000 | ||
Reporter: | Joshua C. Colp (jcolp) | Labels: | pjsip |
Date Opened: | 2018-09-24 12:40:27 | Date Closed: | 2018-09-27 09:21:10 |
Priority: | Major | Regression? | Yes |
Status: | Closed/Complete | Components: | Tests/General |
Versions: | 15.6.0 16.0.0 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | Attachments: | ||
Description: | There is code currently present for attended transfer nominal tests which examines the port for media in the SDP:
{noformat} if self.get_rtp_data(packet) > 10000: # The SDP has an RTP port within Asterisk's range (we know it uses # 10000-20000). {noformat} As of PJSIP 2.8 this is apparently no longer true as it is now using quite a bit higher ports. This breaks the logic and causes the tests to fail: channels.pjsip.transfers.attended_transfer.nominal.callee_local_direct_media channels.pjsip.transfers.attended_transfer.nominal.caller_local_direct_media | ||
Comments: | By: Friendly Automation (friendly-automation) 2018-09-27 09:21:12.748-0500 Change 10259 merged by George Joseph: res_rtp_asterisk: Raise event when RTP port is allocated [https://gerrit.asterisk.org/10259|https://gerrit.asterisk.org/10259] By: Friendly Automation (friendly-automation) 2018-09-27 09:21:25.338-0500 Change 10262 merged by George Joseph: res_rtp_asterisk: Raise event when RTP port is allocated [https://gerrit.asterisk.org/10262|https://gerrit.asterisk.org/10262] By: Friendly Automation (friendly-automation) 2018-09-27 09:21:55.518-0500 Change 10256 merged by George Joseph: res_rtp_asterisk: Raise event when RTP port is allocated [https://gerrit.asterisk.org/10256|https://gerrit.asterisk.org/10256] By: Friendly Automation (friendly-automation) 2018-09-27 09:22:41.002-0500 Change 10258 merged by George Joseph: res_rtp_asterisk: Raise event when RTP port is allocated [https://gerrit.asterisk.org/10258|https://gerrit.asterisk.org/10258] |