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| 09:13.56 | *** topic/#asterisk by file -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.36.0 (2020/09/03) 16.13.0 (2020/09/03) Standard: 17.7.0 (2020/09/03); 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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| 13:10.37 | n0tiz | Hey, once a patch has been reviewed, and is then approved, I assume that's merged in. Jenkins appears to run a build, what should I be doing if that build becomes unstable? |
| 13:15.16 | file | nothing, it will be reviewed to see if the failure is from a flaky test and if so then it will be manually merged |
| 13:17.14 | n0tiz | Cool, thanks again! Just making sure I follow procedures, and don't do anything stupid! |
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| 14:37.15 | schangg | Hello |
| 14:38.36 | schangg | When my asterisk (which is exposed on the Internet) server receives a call from a device that is on an external network, then asterisk tries to send RTP traffic to the internal IP of that device |
| 14:38.58 | schangg | <asterisk>---internet---<router><sip-client> |
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| 14:39.58 | schangg | I see in sip signalling that the sip client is sending 192.xxxx as IP |
| 14:40.12 | schangg | so I am not sure who to blame (asterisk or the qip client ?) |
| 14:40.19 | schangg | s/qip/sip/ |
| 14:49.56 | Samot | Wait, wait, wait... |
| 14:50.11 | Samot | schangg: You have devices coming into the container over the Internet? |
| 14:53.43 | file | if configured to do so Asterisk will latch onto the source of media, and send RTP to it, but that does require it to have received media |
| 14:53.46 | Samot | schangg: I only asked because less than 24 hours ago you told me this was internal only. I asked about SIP trunks, etc. all about that whole double NAT and you said "nope, it's internal" |
| 14:55.09 | schangg | Hi Samot |
| 14:55.13 | schangg | still no trunk |
| 14:55.20 | Samot | But I said the PSTN |
| 14:55.24 | Samot | The Internet |
| 14:55.24 | schangg | now that this works from internal |
| 14:55.32 | Samot | External. |
| 14:55.36 | Samot | My whole things about double NAT. |
| 14:55.42 | schangg | I am port forwarding my public IP to the asterisk host |
| 14:55.48 | schangg | yeah I got that |
| 14:56.15 | schangg | but double NAT or not, the SIP INVITE contains the internal IP of the sip client |
| 14:56.26 | schangg | and asterisk then tries to send rtp traffic to that IP |
| 14:56.48 | Samot | So either the SIP client or the router in front of it is poorly configured. |
| 14:56.51 | schangg | in terms of configuration, I have set externip to the pub IP, localnet and nat=yes |
| 14:57.11 | schangg | all the router does is port forward |
| 14:57.20 | Samot | No, where the SIP client is at |
| 14:57.22 | schangg | nothing like SIP-aware router or such |
| 14:57.30 | Samot | The SIP client is remote correct? |
| 14:57.38 | schangg | you mean the exit router of the sip client ? |
| 14:57.50 | schangg | yeah, it is a cell phone connected to 4G |
| 14:58.00 | Samot | OK so this is over mobile. |
| 14:58.02 | schangg | in 4G the IP phone is NATed by the operator |
| 14:58.19 | Samot | I'm aware. |
| 14:58.36 | schangg | yeah over mobile but from a network perspective it is just like any computer accessing the internet through a router |
| 14:58.40 | Samot | So the sip client is a softphone on a mobile device use 4G data. |
| 14:58.47 | schangg | yes |
| 14:59.08 | schangg | so the sip client is not necessarily the ne to blame ? |
| 14:59.34 | Samot | Sure it can be |
| 14:59.39 | Samot | What is the sip client? |
| 15:00.06 | schangg | you said asterisk if well configured is supposed to look at the actual originating IP instead of what is advertised in the SIP signalling packet |
| 15:00.20 | schangg | mizudroid |
| 15:00.22 | Samot | Did I? |
| 15:00.24 | schangg | just picked one at random |
| 15:00.29 | Samot | I don't recall saying that. |
| 15:00.37 | schangg | Samot no you did not. File did |
| 15:00.42 | schangg | sorry :) |
| 15:01.02 | Samot | You're still using Chan_SIP correct? |
| 15:01.06 | schangg | no |
| 15:01.17 | schangg | using the stuff you mentioned yesterday |
| 15:01.29 | schangg | and after using that it works fine from the internal network |
| 15:01.36 | Samot | Show me the endpoint config for this remote device. |
| 15:01.43 | schangg | however, it is in sip.conf that I have put the nat=yes etc |
| 15:01.53 | schangg | sure |
| 15:02.06 | Samot | If you're not using Chan_SIP, sip.conf is pointless. |
| 15:02.22 | schangg | https://hastebin.com/qilerupese.ini |
| 15:02.39 | schangg | ok then should I just remove the file ? |
| 15:03.45 | file | the given PJSIP configuration includes no NAT options or functionality |
| 15:04.06 | schangg | file correct, I thought these still had to be set in sip.conf |
| 15:04.26 | igcewieling | you were wrong |
| 15:04.38 | schangg | so should I create a [general] section in pjsip.conf and then use the same nat=yes externip= and localnet= ? |
| 15:05.15 | schangg | igcewieling indeed, sorry last time I have used asterisk was like 2010 so I was ignorant of recent changes |
| 15:05.24 | Samot | No. |
| 15:05.34 | Samot | rewrite_contact=yes |
| 15:05.49 | Samot | force_rport=yes |
| 15:05.58 | file | rtp_symmetric=yes |
| 15:06.19 | Samot | Yup. |
| 15:06.29 | Samot | force_rport is default yes but still.. |
| 15:06.46 | Samot | Those three settings per endpoint. |
| 15:06.55 | Samot | Chan_PJSIP doesn't have a global concept. |
| 15:07.09 | Samot | For the endpoint/aor settings. |
| 15:07.27 | schangg | and no more nat=yes as well as externip ? |
| 15:07.38 | Samot | Well you need to show your transports |
| 15:07.49 | Samot | Oh I see it |
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| 15:07.54 | Samot | Yeah that transport is messed up too |
| 15:08.03 | Samot | local_net= |
| 15:08.09 | Samot | external_media_address |
| 15:08.32 | Samot | external_signaling_address |
| 15:10.05 | schangg | https://hastebin.com/vepumugoho.ini |
| 15:10.17 | schangg | would something like this work ? |
| 15:11.50 | Samot | No |
| 15:12.00 | Samot | Because those last three things I mentiond go in the _transport_ |
| 15:12.05 | schangg | sorry, what did I get wrong then |
| 15:12.12 | Samot | Hence me first asking to see the transports. |
| 15:12.12 | schangg | okok |
| 15:14.01 | schangg | it is still sending to the internal IP of the sip client :( |
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| 15:20.43 | Samot | Show a call with a SIP trace... |
| 15:20.47 | Samot | pjsip set logger on |
| 15:20.51 | Samot | Make the call, show it. |
| 15:22.04 | schangg | https://hastebin.com/uxapazafok.makefile |
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| 15:22.48 | file | it is receiving no media, so it can not lock on to the source address |
| 15:23.47 | schangg | file like it does not receives any rtp packet from the phone so it decides it wont send any either ? |
| 15:23.59 | Samot | You restarted Asterisk right? |
| 15:24.08 | schangg | yes |
| 15:24.09 | schangg | https://hastebin.com/kayeqolivo.makefile |
| 15:24.23 | schangg | maybe that capture is better because previous call was for a non existing number |
| 15:24.24 | file | you can't send media to somewhere if you don't know where that somewhere is. |
| 15:24.35 | file | you can only send it to where it told you to send it initially. |
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| 15:24.58 | schangg | file is this happening through signalling or upon receiving rtp packets ? |
| 15:25.19 | schangg | oh ok you mean the SIP message from the sip phone do not contain media information |
| 15:25.28 | file | if "rtp_symmetric" is set to "yes" then upon reception of RTP packets the sending of media will switch to the source of it |
| 15:25.49 | file | the SIP signaling contains SDP which states where media should go, but Asterisk will only use that until the above happens |
| 15:25.51 | schangg | file ok |
| 15:30.27 | schangg | fixed |
| 15:30.47 | schangg | messed up something inmy port forward :( |
| 15:31.10 | schangg | all working well ! |
| 15:38.20 | schangg | Thanks a lot for your precious help !!!! :) |
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| 17:08.03 | acro458 | what is the difference between PCMA and G711a? |
| 17:08.27 | acro458 | are these two names for the same thing? |
| 17:09.28 | acro458 | (similar to G711u vs PCMU)? |
| 17:09.31 | igcewieling | The first one is a stupid name for the second. |
| 17:10.58 | acro458 | ok thanks |
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| 17:27.03 | drmessano | Dumber than the ULAW/ALAW convention? |
| 17:28.31 | pramsky | not compared to those |
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| 19:02.40 | igcewieling | goes back to judging books by their covers. |
| 19:06.21 | erichowey | everyone just needs to use opus :P |
| 19:08.00 | sibiria | if we're lucky most cellular operators will finally have bumped up to G.711 in a decade. or two. |
| 19:08.33 | sibiria | G.722* |
| 19:09.00 | igcewieling | T-Mobile appears to use G722 (or other HD codec) for on-net calls between phones which support it. |
| 19:09.10 | erichowey | yeah, which is better. here in the US, they wont traverse G.722 outside of the mobile networks though. Any handoff to a voip connection is downgraded to G.711 |
| 19:09.19 | erichowey | pretty crappy |
| 19:09.38 | sibiria | a lot of euro operators support G.722 now, but there's still a long way to go |
| 19:09.50 | erichowey | i had a T-Mobile -> AT&T Wireless call use G.722 which amazed me |
| 19:09.54 | igcewieling | One of the few co-workers who are not afraid to call me has T-mobile and I get VERY good audio quality and "HD" indicator. |
| 19:09.55 | erichowey | for sure |
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