| Summary: | ASTERISK-17109: Codec negotiation algorithm | ||
| Reporter: | Marco Marzetti (manzo_zeti) | Labels: | |
| Date Opened: | 2010-12-15 07:19:26.000-0600 | Date Closed: | 2011-06-07 14:01:02 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Codecs/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | According to the provided documentation ( sip.conf ), when Asterisk is placing a call the CODEC used will be the first CODEC in the allowed CODEC's that the caller indicates that it supports. So if, for instance, we have the following setup: [caller] disallow=all allow=g729 allow=alaw [called] disallow=all allow=alaw allow=gsm RTP stream from caller to called would be transcoded from $WATHEVER to G.729. Such behavior is sub-optimal when $CUSTOMERS devices come in the scene since we can't be sure that our configured codec preferences are the same as theirs. I think Asterisk should place call using the first CODEC sent by the caller in the INVITE message that [caller] indicates that it supports. | ||
| Comments: | By: Igor Zamocky (dzajro) 2010-12-15 11:33:05.000-0600 There are many scenarios benefiting from current mechanism. But maybe it could be configuration related. Something like "honor host preferences" vs. "honor my preferences". By: Paul Belanger (pabelanger) 2010-12-15 14:13:28.000-0600 Features requests are no longer submitted to or accepted through the issue tracker. Features requests are openly discussed on the mailing lists [1] and Asterisk IRC channels and made note of by Bug Marshals. [1] http://www.asterisk.org/support/mailing-lists | ||