[asterisk-users] Asterisk 14 audio quality with remote files
Hello everyone, I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutting the audio frequencies above 4Khz. The call is established with G.722 and the audio file is mono 16Khz 16 bit sln16 extension. What can I do to improve the sound quality? Is there any way to not have asterisk cut the audio frequencies? best regards Tiago -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pjsip: asterisk can't decide which codec to use
Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extension -> asterisk -> provider A -> provider B -> asterisk. Asterisk initially sends invites using g722 and g711 and gets exactly this invite back as incoming call. The answer is g722,g711 in the ok sdp. Now, Asterisk can't decide, which codec to use. It frequently changes the codec just as it likes to apparently without any visible reason. [2017-05-11 17:28:03] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:03] DEBUG[5113][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:04] DEBUG[5123][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:04] DEBUG[5113][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:04] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:04] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:13] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:13] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:19] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:19] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:23] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:23] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:23] DEBUG[5123][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:23] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:28] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:28] DEBUG[5123][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw 003a -> inbound channel (callee) 0039 -> outbound channel (caller) If I'm doing exactly the same call originated with another extension, there can't be seen these frequent changes. But the strange thing is, that in both cases the part between extension and asterisk doesn't show any codec changes ... . Deeper investigations show, that if the conference (callee) sends the first rtp package (-> g711 - should be g722), things are going choppy, if the extension (caller) sends the first package (g722), things are running stable. Any idea to convince asterisk always to use the first codec of ok sdp or how to convince asterisk to put only one codec to ok sdp (the first). Thanks, regards, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip: asterisk can't decide which codec to use
On 05/12/2017 at 07:46 PM, Michael Maier wrote: Forgot to mention: It's actual asterisk 13 branch from today (version before I tested, which has the same problem, was 13.15). Regards, Michael Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extension -> asterisk -> provider A -> provider B -> asterisk. Asterisk initially sends invites using g722 and g711 and gets exactly this invite back as incoming call. The answer is g722,g711 in the ok sdp. Now, Asterisk can't decide, which codec to use. It frequently changes the codec just as it likes to apparently without any visible reason. [2017-05-11 17:28:03] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:03] DEBUG[5113][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:04] DEBUG[5123][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:04] DEBUG[5113][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:04] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:04] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:13] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:13] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:19] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:19] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:23] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:23] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:23] DEBUG[5123][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:23] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722 [2017-05-11 17:28:28] DEBUG[5121][C-003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw [2017-05-11 17:28:28] DEBUG[5123][C-0039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw 003a -> inbound channel (callee) 0039 -> outbound channel (caller) If I'm doing exactly the same call originated with another extension, there can't be seen these frequent changes. But the strange thing is, that in both cases the part between extension and asterisk doesn't show any codec changes ... . Deeper investigations show, that if the conference (callee) sends the first rtp package (-> g711 - should be g722), things are going choppy, if the extension (caller) sends the first package (g722), things are running stable. Any idea to convince asterisk always to use the first codec of ok sdp or how to convince asterisk to put only one codec to ok sdp (the first). Thanks, regards, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip: asterisk can't decide which codec to use
On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote: > > If I'm doing exactly the same call originated with another extension, > there can't be seen these frequent changes. But the strange thing is, > that in both cases the part between extension and asterisk doesn't show > any codec changes ... . > > Deeper investigations show, that if the conference (callee) sends the > first rtp package (-> g711 - should be g722), things are going choppy, > if the extension (caller) sends the first package (g722), things are > running stable. > > > Any idea to convince asterisk always to use the first codec of ok sdp > or how to convince asterisk to put only one codec to ok sdp (the first). This is not currently an option in chan_pjsip but I'd suggest filing an issue[1] for this scenario with all available information. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pjsip: asterisk can't decide which codec to use
On 05/12/2017 at 08:49 PM, Joshua Colp wrote: > On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote: > > > >> >> If I'm doing exactly the same call originated with another extension, >> there can't be seen these frequent changes. But the strange thing is, >> that in both cases the part between extension and asterisk doesn't show >> any codec changes ... . >> >> Deeper investigations show, that if the conference (callee) sends the >> first rtp package (-> g711 - should be g722), things are going choppy, >> if the extension (caller) sends the first package (g722), things are >> running stable. >> >> >> Any idea to convince asterisk always to use the first codec of ok sdp >> or how to convince asterisk to put only one codec to ok sdp (the first). > > This is not currently an option in chan_pjsip but I'd suggest filing an > issue[1] for this scenario with all available information. > > [1] https://issues.asterisk.org/jira https://issues.asterisk.org/jira/browse/ASTERISK-26996 Thanks, regards, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
