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| 01:45.09 | Tagor | how can I set a global var as soon as dial() was answered? |
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| 10:43.07 | oolivero45 | Hi everyone, sorry if this is the wrong place to ask, but I have a question. Are you able to get Asterisk to re-broadcast the video from a SIP call as a separate h264 video stream over the local network? |
| 10:43.20 | oolivero45 | I've got a client which can only receive audio over SIP, but not video, but it can receive a h264 stream over the network. I want to try separating the video from the call into its own stream for that client to receive, but can't work out how/if that can be done. |
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| 12:44.06 | Dovid | As per the RFC in a SDP which IP do we look at for RTP is it the onwer/creater or the connection information? |
| 12:47.18 | file | the connection information is the IP where media is to be sent |
| 12:47.55 | Kobaz | oh yeah |
| 12:47.59 | Kobaz | i had a question last night |
| 12:48.18 | Kobaz | https://community.asterisk.org/t/configure-multiple-transport-for-per-endpoint/70989/6 |
| 12:48.23 | Kobaz | So..... apparently you can leave transport blank on an endpoint?? This doesn't appear to work |
| 12:48.33 | file | you haven't defined how it doesn't work |
| 12:48.45 | Kobaz | oh, sure |
| 12:48.57 | Kobaz | Like, if endpoints are registering to you over multiple transports... they all go unreachable if you take out transport= |
| 12:49.13 | Kobaz | then, pjsip has no idea how to route to them |
| 12:49.17 | file | haven't seen that before. |
| 12:49.22 | Kobaz | asterisk 16.10 |
| 12:49.29 | Kobaz | I'll try 16.14 |
| 12:49.47 | Kobaz | and once you put transport=udp, they all come back |
| 12:50.09 | Kobaz | i have two udp transports on two local ip addresses (bound to two different publics) |
| 12:50.11 | file | it does work, but if it's not working then it's likely something dependent on your specific environment or configuration |
| 12:50.33 | Kobaz | so not defining a transport entirely, should auto-detect the transport? |
| 12:50.58 | file | it will choose a transport to use based on the protocol specified in the URI and will try to consult the local routing table to find an appropriate one based on that |
| 12:51.28 | Kobaz | and that's when transport is undefined? |
| 12:51.33 | file | yes. |
| 12:51.36 | Kobaz | hmm |
| 12:51.38 | Dovid | file: I am having an issue with chan_sip on 11.25. Seems Asterisk is locking in on src IP + Port and not looking at the sdp. AFAIK Pjsip was introduced in 12. Is that correct? |
| 12:52.16 | file | yes, however both chan_sip and chan_pjsip will behave that way if NAT options are enabled. |
| 12:52.23 | Kobaz | Dovid: I've found many lockups/crashes with 11.20 - 11.25... join the club... unfortunately |
| 12:52.38 | Kobaz | oh locking in/onto... not locking up |
| 12:53.22 | Kobaz | Dovid: you have nat options enabled, then if will not trust the SDP... correct |
| 12:53.40 | Dovid | kobaz: nat=no |
| 12:54.05 | Kobaz | what's your packet look like? |
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| 13:17.25 | Dovid | kobaz: my invite or the carriers SDP back to me? |
| 13:26.56 | Kobaz | the whole thing |
| 13:27.02 | Kobaz | actually... the more the merrier |
| 13:27.20 | Kobaz | start the finish the whole sip conversation |
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