Summary:	ASTERISK-25845: res_pjsip_sdp_rtp: Wrong audio codec used when video enabled
Reporter:	Beytullah ARSLAN (barslan2)	Labels:
Date Opened:	2016-03-15 07:30:06	Date Closed:
Priority:	Minor	Regression?	No
Status:	Open/New	Components:	Resources/res_pjsip_sdp_rtp
Versions:	13.7.2	Frequency of Occurrence	Frequent
Related Issues:
Environment:	Gento-Linux, 4.1.12 Asterisk compiled directly from source with pjsip 2.4.5 support	Attachments:	( 0) pjsipsetloggeron.txt
Description:	The codec is negotiated with the call initiator when the initator places a call and a codec is choosed. For instance g722 was choosed between the caller and Asterisk. Between called party and Asterisk a suitable another codec for instance ulaw was choosed. When the sound starts from the caller, for the translation, Asterisk looks for the list of available codecs of the called party's allowed codec list and if it founds the caller's codec (g722) it uses it. INSTEAD of using the negotiated codec (ulaw) af called party. And this a problem that can be simulated easily with different codecs also. And I think you can fix this problem easily.
Comments:	By: Asterisk Team (asteriskteam) 2016-03-15 07:30:07.728-0500 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. 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: Joshua C. Colp (jcolp) 2016-03-17 08:58:51.515-0500 Please attach the configuration for the endpoints as well. By: Asterisk Team (asteriskteam) 2016-03-31 12:00:01.863-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: Beytullah ARSLAN (barslan2) 2016-04-01 01:13:53.828-0500 {noformat} [yerel-telefon-sablonu](!) type=endpoint direct_media=yes disable_direct_media_on_nat=yes message_context=mesajlar disallow=all allow=g722 allow=g729 allow=ulaw allow=alaw allow=h264 allow_subscribe=yes sub_min_expiry=30 transport=transport-udp [auth-userpass](!) type=auth auth_type=userpass [aor-single-reg](!) type=aor remove_existing=yes     ; Determines whether new contacts replace existing ones max_contacts=2 qualify_frequency=5 authenticate_qualify=yes ;===== [4160](yerel-telefon-sablonu) context=custom-uluslararasi auth=4160 aors=4160 callerid="caller id" <4160> call_group= 10 pickup_group= 10 allow_subscribe=yes sub_min_expiry=30 [4160](auth-userpass) username=4160 password=pass [4160](aor-single-reg) ;===== {noformat} By: Asterisk Team (asteriskteam) 2016-04-01 01:13:54.045-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable.