| Summary: | ASTERISK-25684: codec: Translation of slin16 results in noise | ||||
| Reporter: | Victor Sverdlin (victor_sverdlin) | Labels: | |||
| Date Opened: | 2016-01-11 08:22:20.000-0600 | Date Closed: | 2016-01-26 08:00:19.000-0600 | ||
| Priority: | Minor | Regression? | |||
| Status: | Closed/Complete | Components: | Codecs/General Resources/res_rtp_asterisk | ||
| Versions: | 13.1.0 13.4.0 13.6.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Ubuntu 14.4 i686 3.19.0-42 on VMware WS 10 FreeBSD 10.2 i386 on VirtualBox 5.0.10 Ubuntu 14.4 i686 3.19.0 on bare hw (intel core i5) | Attachments: | ( 0) cli_output.txt ( 1) myDebugLog.txt ( 2) slin_noise_01.pcap ( 3) slin_noise_15jan2015.pcap | ||
| Description: | Then one peer is slin (L16/8000) or slin16 (L16/16000) and second peer is other codec (tested with ulaw, gsm and speex) slin peer listen only noise. Noise disappear if other peer muted. Slin peer can listen voice if signal is gained down by AGC set to 10 and less (anyway sound is distorted).
Both peers use sip channel. Testes with several peers: - slin: MicroSIP 3.10.9, custom HW device - other: antiSIP 4.2.9 (Android), SFLphone 1.3.0, Zoiper 3.3.25608 Asterisk compiled with gcc 4.8.5(FreeBSD) and 4.8.4(Ubuntu). On Ubuntu Asterisk compiled with DONT_OPTIMIZE option. RTP capture will be attached. | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2016-01-11 08:22:22.198-0600 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) 2016-01-13 16:24:53.692-0600 What byte order is the signed linear being sent in? If this is wrong then you would get exactly what you are hearing. By: Victor Sverdlin (victor_sverdlin) 2016-01-14 03:35:57.968-0600 Hi, Joshua, 1. I listen noise on the stream from asterisk to slin16 peer. Stream sent from the slin16 to asterisk is correctly converted to ulaw. 2. Byte order for slin16 (L16) on RTL is always BE (MSB first). Then two L16 peers are connected directly (via native channel) both streams have no noise. 3. Look in the attached pcap: for instance, compare frames 84 (asterisk to microsip) and 85 (microsip to asterisk). Frame 84 has FF in most of MSBs, but 84 has 00. Payloads have the same codec (L16). With best regards, Victor By: Rusty Newton (rnewton) 2016-01-14 07:11:08.382-0600 It may be unnecessary, but we like to have all the info we can up front. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk during a call. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Joshua C. Colp (jcolp) 2016-01-26 08:00:19.875-0600 After examining closer and looking at things this appears to be a duplicate of the older ASTERISK-24858. Closing this out as a result. | ||||