| Summary: | ASTERISK-27675: Feature Request: Single Codec in Response | ||
| Reporter: | Carsten Bock (CarstenBock) | Labels: | pjsip |
| Date Opened: | 2018-02-14 10:37:10.000-0600 | Date Closed: | 2018-02-14 10:57:49.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_pjsip |
| Versions: | 13.15.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Debian 9 | Attachments: | |
| Description: | Hi,
as you may know, we love IMS. And we love using Asterisk (PJSIP) with IMS, as it makes our life easier... :-) Chan_pjsip does a great job! However, we have a compatibility issue: In our Lab-Setup, we use Asterisk as a B2BUA between VoLTE handsets and the "rest of the world". The VoLTE handsets usually speak AMR-Codecs, while in the rest of the world we use G711/G722. So Asterisk needs to do Transcoding, which it does great. In our setup we have: disallow=all allow=g722 allow=amr allow=amrwb allow=ulaw allow=alaw Call comes in (with AMR / AMRWB) and goes out with G722, AMR, AMRWB, G711. Reply comes in as G722, and reply goes out with AMR and AMRWB. While I believe it is technically correct to reply with both codecs, it confuses our Lab-Samsung Handsets, so we simply get "no audio" instead of anything else. If I comment either AMR or AMRWB, the reply contains only one codec and the phone is happy. It would be desirable, to have an option to always only have one codec in the reply instead of all compatible ones. Of course we've raised the issue with Samsung, but it takes them ages to fix this. Thanks, Carsten | ||
| Comments: | By: Asterisk Team (asteriskteam) 2018-02-14 10:37:11.283-0600 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. 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]. By: Richard Mudgett (rmudgett) 2018-02-14 10:56:10.147-0600 Features requests without patches are not accepted through the issue tracker. Features requests are openly discussed on the mailing lists, forums, and IRC [1]. Please see the Asterisk Issue Guidelines [2] for more information on feature request and patch submission. [1] http://asterisk.org/community/discuss [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines The asymmetric_rtp_codec option was added to do just this in v13.17.0. By: Kristijan Vrban (vrban) 2018-02-15 04:16:09.566-0600 Shoud be doable with PJSIP_MEDIA_OFFER? Call from handsets with AMR AND AMRWB, then set AMRWB via PJSIP_MEDIA_OFFER. Call with only AMR OR AMRWB, set AMR OR AMRWB via PJSIP_MEDIA_OFFER. Not tested, but theoretical this is what you are looking for? | ||