| Summary: | ASTERISK-27702: allow video/h264 breaks symmetric rtp | ||
| Reporter: | Jørgen H (jorgen) | Labels: | pjsip |
| Date Opened: | 2018-02-27 04:25:03.000-0600 | Date Closed: | 2018-02-28 11:13:50.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Resources/res_pjsip |
| Versions: | 14.5.0 15.2.2 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | linux x64 Verified on asterisk 15.2.2 with pjsip 2.7.2 asterisk 14.5 and pjsip 2.6 | Attachments: | |
| Description: | Adding codec h264 in allow-list result in that rtp_symmetric gets ignored and asterisk sends rtp audio to the ip-address in the sdp even when h264 is not in use.
Config: rewrite_contact = yes rtp_symmetric = yes force_rport = yes direct_media = no allow = !all,opus,g722,alaw,ulaw,h264 SDP from answering telephone when audio works without h264 in allow list: v=0 o=MxSIP 0 1 IN IP4 10.24.5.115 s=SIP Call c=IN IP4 10.24.5.115 t=0 0 m=audio 3000 RTP/AVP 9 8 0 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv SDP when h264 is enabled and audio does not work (audio is sent to 10.24.5.115): v=0 o=MxSIP 0 1 IN IP4 10.24.5.115 s=SIP Call c=IN IP4 10.24.5.115 t=0 0 m=audio 3000 RTP/AVP 9 8 0 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv m=video 0 RTP/AVP 99 | ||
| Comments: | By: Asterisk Team (asteriskteam) 2018-02-27 04:25:04.381-0600 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Joshua C. Colp (jcolp) 2018-02-27 04:49:36.609-0600 You'll need to provide the "rtp set debug on" output as well to show media actually arriving and to show where we are sending it. By: Jørgen H (jorgen) 2018-02-28 10:33:40.280-0600 with h264: Sent RTP packet to 10.24.5.115:3000 (type 08, seq 006566, ts 1173570704, len 000160) and obviously no got rtp packet from that ip. Without h264 it is sent to correct ip and received from correct ip. By: Joshua C. Colp (jcolp) 2018-02-28 10:37:49.761-0600 Please provide a complete attachment. If Asterisk isn't receiving media then it can't lock on to the source address and send media there. By: Jørgen H (jorgen) 2018-02-28 11:10:42.299-0600 Ah. Yes, the phone doesn't send any rtp to any ip-address. I thought asterisk was supposed to send rtp to the ip regardless if the phone sent anything. Firmware upgrade of phone solved the problem. Sorry about the incorrect report (; By: Joshua C. Colp (jcolp) 2018-02-28 11:13:50.273-0600 Asterisk will send media to the source only when it gets some. It can't deduce what the address will be until it sees the media. | ||