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| 05:42.18 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.0.1, 13.37.1, 16.14.1 (2020/11/05) Standard: 17.8.1 (2020/11/05); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 13:48.36 | AL13N_lappy | codec negotiation is annoying... |
| 13:50.17 | AL13N_lappy | EXT201 => AST (opus,alaw) , AST => siptrunk (alaw,ulaw) ; siptrunk => AST (alaw) ; AST => EXT201 (opus,alaw,ulaw) ... |
| 13:50.59 | AL13N_lappy | the last one should be (alaw), cause that's what remote is set... |
| 13:53.29 | Samot | No it shouldn't be. |
| 13:53.47 | Samot | Asterisk is a Back2Back User Agent. |
| 13:54.10 | AL13N_lappy | https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation ( i set in all 4 ; prevent, intersect and pending) in order to avoid trouble... |
| 13:54.27 | Samot | OK.. |
| 13:54.41 | Samot | That doesn't change the fact it's a B2BUA |
| 13:54.48 | Samot | So you have two ASterisk systems? |
| 13:54.50 | AL13N_lappy | ok |
| 13:55.07 | AL13N_lappy | well, i supose the upstream sip trunk is likely asterisk |
| 13:55.28 | Samot | Pardon? |
| 13:55.42 | Samot | So the SIP trunks are from a third party? |
| 13:55.45 | file | advanced codec negotiation is not yet complete, the foundational work is present but the PJSIP work will be available in a future release, there is limited new configuration available now |
| 13:55.56 | AL13N_lappy | ic |
| 13:56.22 | Samot | Either way, it still may not matter. |
| 13:56.33 | AL13N_lappy | Samot: wdym with B2B User Agent? how does this change the codec negotiation? |
| 13:56.35 | Samot | AL13N_lappy: Are the two SIP trunks from a provider? |
| 13:56.47 | AL13N_lappy | Samot: no, |
| 13:56.53 | Samot | AL13N_lappy: EXT201 => AST |
| 13:56.57 | Samot | That's channel 1 |
| 13:57.05 | AL13N_lappy | ok, so my laptop is EXT201 ; the AST is my asterisk server |
| 13:57.12 | Samot | The codecs will be determined between that device and Asterisk. |
| 13:57.13 | AL13N_lappy | sip trunk is the upstream sip trunk |
| 13:57.24 | Samot | Who is the UPSTREAM? |
| 13:57.32 | AL13N_lappy | a telephone company |
| 13:57.39 | Samot | So a third party |
| 13:57.42 | AL13N_lappy | yes |
| 13:57.51 | Samot | I asked and you said no. |
| 13:58.05 | AL13N_lappy | ah, no, you said both 3rd parties, but there's only one |
| 13:58.12 | Samot | Sigh. |
| 13:58.15 | AL13N_lappy | sorry |
| 13:58.47 | Samot | OK are there two Asterisk systems? |
| 13:58.59 | AL13N_lappy | maybe |
| 13:59.04 | AL13N_lappy | Samot: wait, ok |
| 13:59.15 | Samot | 8:50:15 AM <AL13N_lappy> EXT201 => AST (opus,alaw) , AST => siptrunk (alaw,ulaw) ; siptrunk => AST (alaw) ; AST => EXT201 (opus,alaw,ulaw) ... |
| 13:59.24 | Samot | How the hell does that lay out properly? |
| 13:59.28 | AL13N_lappy | Samot: EXT201 (my laptop), AST( my asterisk), SIP (3rd party telephone company) |
| 13:59.49 | Samot | What you posted looks like two Asterisk systems. |
| 13:59.55 | Samot | 8:50:15 AM <AL13N_lappy> EXT201 => AST (opus,alaw) , AST => siptrunk (alaw,ulaw) ; siptrunk => AST (alaw) ; AST => EXT201 (opus,alaw,ulaw) ... |
| 14:00.00 | Samot | Explain that ^^^^ |
| 14:00.05 | AL13N_lappy | ok |
| 14:00.30 | AL13N_lappy | Samot: ok, so about codec negotiation (i'm looking at the SIP traffic with tcpdump) |
| 14:00.39 | Samot | No. |
| 14:00.42 | AL13N_lappy | EXT201 dials a cellphone |
| 14:00.47 | Samot | Explain that layout |
| 14:00.57 | Samot | Because it looks like two Asterisk systems calling each other through the PSTN |
| 14:01.04 | AL13N_lappy | Samot: that line was not layout, that was the codec traffic on SIP |
| 14:01.13 | Samot | OK |
| 14:01.18 | Samot | So let me try this a gain. |
| 14:01.26 | AL13N_lappy | the layout is like what's up there: Samot: EXT201 (my laptop), AST( my asterisk), SIP (3rd party telephone company) |
| 14:01.36 | Samot | EXT201 calls Asterisk, the codecs will be determined between the device for EXT201 and Asterisk. |
| 14:01.41 | AL13N_lappy | yes |
| 14:01.42 | Samot | Just THAT CHANNEL |
| 14:01.48 | AL13N_lappy | yes |
| 14:02.10 | Samot | If it decides to call the PSTN then a NEW channel is created from AST to the Provider. |
| 14:02.17 | AL13N_lappy | right |
| 14:02.18 | Samot | THAT channel figures out its codecs |
| 14:02.22 | AL13N_lappy | yes |
| 14:02.32 | Samot | Then Asterisk will merge to the two together and transcode if needed. |
| 14:02.43 | AL13N_lappy | ic |
| 14:03.17 | Samot | So EXT201 could end up being Opus between the device and Asterisk but the trunk could be ulaw. |
| 14:03.31 | Samot | So Asterisk would have to transcode Opus<>ulaw |
| 14:03.49 | AL13N_lappy | but, when i look at sip traffic and/or logs, it looks like the first step does not actually has decided a single codec yet, until actually the 2nd channel has decided codec |
| 14:03.59 | AL13N_lappy | Samot: i understand what you say |
| 14:04.09 | Samot | Show the logs. |
| 14:05.10 | file | it doesn't work like you're expecting yet. |
| 14:05.19 | file | it can in the future. |
| 14:06.19 | AL13N_lappy | Samot: https://pastebin.com/PwVkhaJf this is sip traffic, (ast logs to come) |
| 14:06.23 | AL13N_lappy | file: ic |
| 14:06.56 | AL13N_lappy | file: what about the ;incoming_call_offer_pref=remote & ;outgoing_call_offer_pref=local ? |
| 14:07.27 | AL13N_lappy | file: that & ACN looks like it's set up to do something about this |
| 14:07.28 | file | they exist, but that is only one side of the equation |
| 14:07.32 | AL13N_lappy | file: right |
| 14:07.53 | AL13N_lappy | Samot: i guess it's no use then to show logs |
| 14:08.19 | Samot | So what's the problem here? |
| 14:08.21 | AL13N_lappy | file: i thought if i set this on each pjsip endpoint, i could do something about it |
| 14:08.24 | Samot | I see this connect at ulaw |
| 14:08.46 | AL13N_lappy | Samot: the problem is, there is no transcoding between opus and anyting else |
| 14:09.01 | AL13N_lappy | and i'd like to avoid it in any case even if it were possible |
| 14:09.15 | Samot | So then the device shouldn't use opus as one of it's first codecs. |
| 14:09.26 | Samot | Order does matter. |
| 14:10.00 | AL13N_lappy | Samot: well, i want EXT201 -> AST -> EXT202 to use opus; while EXT201 -> AST -> SIPTRUNK to use alaw (like siptrunk wants) |
| 14:10.23 | AL13N_lappy | Samot: i could just have everything use alaw and everything would work without transcoding |
| 14:10.30 | Samot | Again, order matters. |
| 14:11.29 | AL13N_lappy | Samot: yeah, it does |
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| 14:15.48 | AL13N_lappy | file: so, i'm building with upstream pjsip; (version 18.0.0) but i suppose there's some trial patches for this? and is this upstream or planned for the bundled pjsip? cause i'm assuming changes are not in asterisk, but need to be in pjsip? |
| 14:17.59 | AL13N_lappy | i guess i'll just force everything to be alaw atm |
| 14:18.36 | file | there are no PJSIP upstream changes |
| 14:18.38 | file | it's Asterisk |
| 14:19.05 | file | there are no "trial patches" |
| 14:20.21 | sibiria | there's only trial by fire |
| 14:20.33 | sibiria | ** ELECTRIC GUITAR RIFF ** |
| 14:27.03 | Samot | AL13N_lappy: Have you tried PJSIP_MEDIA_OFFER? |
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