| Summary: | ASTERISK-19598: Garbled audio using Page app and MulticastRTP channel | ||||
| Reporter: | Remi Quezada (remiq) | Labels: | |||
| Date Opened: | 2012-03-28 11:58:17 | Date Closed: | 2017-12-14 10:12:16.000-0600 | ||
| Priority: | Minor | Regression? | No | ||
| Status: | Closed/Complete | Components: | Applications/app_page Channels/chan_multicast_rtp | ||
| Versions: | SVN | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Attachments: | ( 0) rtppage.cpt ( 1) rtp-page-capture.txt | |||
| Description: | Getting garbled audio with Multicast RTP and Page application. Multicast RTP works fine with Dial application. I am using the following phones for multicast rtp, all have the same garble audio: Cisco spa504G Cisco spa303 SNOM 821 SIP/256-eng is a Polycom Soundpoint 331 Able to reproduce with dialplan listed below. I also attached cli debug and an IP capture. | ||||
| Comments: | By: Remi Quezada (remiq) 2012-03-28 13:08:33.435-0500 Asterisk is sending the RTP on 209.191.39.117:34560 and I have a Adtran 908 router configured to change 209.191.39.117:34560 to 224.168.168.168:34560. All the Cisco and SNOM phones are configured to receive the multicast RTP on 224.168.168.168:34560. By: Vitaliy Aleksandrov (vitalik) 2012-11-07 04:04:29.478-0600 The problem is still present. I have tested MulticastRTP channel with Dial command and it works really great. But when i'm trying to use it with app_page (that uses confbridge) i'm getting a very garbled audio. Confbridge with SIP channels works great too. Is there any way to send multicast stream to more that one interface without app_page ? p.s. all tests were made at asterisk-11 By: Sean Bright (seanbright) 2017-12-14 10:12:17.103-0600 If you can reproduce this with Asterisk 13.18+, please feel free to re-open. | ||||