Sorry, I've requested for "Local active SDP Session" but I need "sdp.cpp:391] SDP: Local SDP Session" (the ones before the negotiation)
----- Le 15 Mar 16, à 10:21, Josh Nijenhuis j...@nijenhuis.ca a écrit : > m=audio 21018 RTP/AVP 104 0 8 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:104 opus/48000/2 > a=rtpmap:101 telephone-event/8000 > > Well that info helped along with the rest of the output. > > So I got it working in this condition, > Turned opus off in Asterisk, and uncheck opus codec in Ring > m=audio 21448 RTP/AVP 0 8 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > > > Doesn't work in this condition > Turned opus off in Asterisk, check opus codec in Ring > m=audio 17762 RTP/AVP 104 0 8 9 101 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:104 opus/48000/2 > a=rtpmap:101 telephone-event/8000 > > I never changed the codec order as listed in attachment. Just uncheck/check > > could this be related to https://tuleap.ring.cx/plugins/tracker/?aid=113 > > Thanks, > > Josh > On 03/15/16 09:59, Guillaume Roguez wrote: >> Hi, >> >> Weird, just tested right now (but on a Fedora 23) and works fine. >> >> Could you run the daemon (dring) using a console and give it arguments -c -d >> ? >> like this: >> >> /usr/lib/x86_64-linux-gnu/dring -c -d >> >> Then run the Gnome client as usual (be sure to have killed all instances of >> client/daemon processes before) >> >> Try again and search in the log a line saying something like: >> >> [1458049858.284| 4088|sipvoiplink.cpp:929 ] Local active SDP Session: >> >> There are many lines after giving the SDP sent to the asterisk server. >> Look at the one starting by "m=audio" : please, could you past from this one >> and >> all ones after starting with "a=rtpmap"? >> >> Note: not other ones, you can leak your IP :-) >> >> Thanks, >> Guillaume >> >> ----- Le 15 Mar 16, à 9:31, Josh Nijenhuis j...@nijenhuis.ca a écrit : >> >>> Good Morning All, >>> >>> I have successfully compiled >>> ring-daemon >>> ring-lrc >>> ring-client-gnome >>> >>> on gentoo 64 bit linux. >>> >>> All is working well except outgoing codec. >>> >>> When incoming from asterisk server, codec used is ulaw, this works as >>> expected. even though ulaw(g711) not in Ring codec list. >>> >>> Outgoing calls from Ring, codec used is opus, but since asterisk only >>> supports opus pass-through, and phone doesn't support opus, no audio. >>> >>> Codec list in Ring in order displayed: >>> G722 >>> PCMA >>> PCMU >>> opus >>> >>> when I disable opus and leave other 3 checked. I get call failed >>> immediately, didn't even reach asterisk... >>> >>> So my question is, is there a wiki page or info on how Ring uses codecs >>> or compiles against them? >>> >>> very weird that ulaw will work incoming but its not displayed in the >>> codecs list... >>> >>> Thanks, >>> >>> Josh >>> >>> >>> _______________________________________________ >>> Ring mailing list >>> Ring@lists.savoirfairelinux.net > >> https://lists.savoirfairelinux.net/mailman/listinfo/ring _______________________________________________ Ring mailing list Ring@lists.savoirfairelinux.net https://lists.savoirfairelinux.net/mailman/listinfo/ring
