Summary: | ASTERISK-16379: CODEX SPEEX 16K | ||
Reporter: | celya (celya) | Labels: | |
Date Opened: | 2010-07-16 03:03:16 | Date Closed: | 2014-07-01 15:40:48 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Codecs/codec_speex |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) Speex16-2 ( 1) Speex16-wireshark.pcap | |
Description: | With the asterisk trunk, 16KHz speex codecs have a poor quality. It's sounds like a frequency problem. If you use SFLphone 0.9.8, when you establish a communication you have on your screen "SPEEX/8000" | ||
Comments: | By: Leif Madsen (lmadsen) 2010-07-16 10:36:44 Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. You may find it helpful to read the Asterisk Issue Guidelines http://www.asterisk.org/developers/bug-guidelines. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. the specific steps or actions you took that caused you to encounter the problem, 2. the behavior you expected, and 3. the behavior you actually encountered (in as much detail as possible). This likely includes output from the console with debug level logging, a SIP trace (if this is SIP related), and configuration information such as dialplan (e.g. extensions.conf) and channel configuration (e.g. sip.conf). Thanks! By: celya (celya) 2010-07-16 11:45:03 My SIP account : [34479] type=friend username=34479 insecure=very qualify=no nat=yes host=dynamic canreinvite=no context=bbb-voip disallow=all allow=speex16 password=toto In my context I juste use Playback(conf-placeintoconf) The SDP is write : sip show channel 209b23ea-0e8c-42fe-915b-2446aa2adb14 * SIP Call Curr. trans. direction: Incoming Call-ID: 209b23ea-0e8c-42fe-915b-2446aa2adb14 Owner channel ID: SIP/34479-0000000a Our Codec Capability: 0x200000000 (speex16) Non-Codec Capability (DTMF): 1 Their Codec Capability: 0x200000000 (speex16) Joint Codec Capability: 0x200000000 (speex16) Format: 0x200000000 (speex16) T.38 support No Video support No MaxCallBR: 384 kbps Theoretical Address: 192.168.1.128:5060 Received Address: 192.168.1.128:5060 SIP Transfer mode: open Force rport: Yes Audio IP: 192.168.1.70 (local) Our Tag: as42ab2675 Their Tag: e0ed6757-8519-46fa-8fb7-f6670cf7cc62 SIP User agent: Username: 34479 Peername: 34479 Original uri: sip:34479@192.168.1.128:5060 Caller-ID: 34479 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: sip:34479@192.168.1.128:5060 DTMF Mode: rfc2833 SIP Options: 100rel replaces replace Session-Timer: Inactive I try to change the SPEEX parameters in codecs.conf, but it's always the same problem. If I have two SIP account with SPEEX16 codecs, and juste a dial between us, it the same problem, the sound is awful. You can find my wireshark capture. By: Leif Madsen (lmadsen) 2010-07-20 10:34:30 I'd suggest you try another softphone as well to determine that it isn't a localized bug in the phone you're using. By: Leif Madsen (lmadsen) 2010-07-20 10:35:18 Please upload text files with the extension .txt or the extension .pcap if it is a wireshark trace. By: celya (celya) 2010-07-20 12:15:43 I tried with xlite under windows, exactly the same poor quality in 16Khz. The 8Hkz work fine on both softphone and OS. I upload the pcap file. By: Timo Teräs (fabled) 2010-08-07 04:24:04 What is the source format of conf-placeintoconf? Have you installed 16khz speaks? I have similar setup here and the voice sounds just fine with SFLphone 0.9.8. I do see speex/8000 on display, but it is actually speex/16000. It sounds like SFLphone bug that it shows the selected codec wrong. By: Malcolm Davenport (mdavenport) 2014-06-17 14:54:27.132-0500 Do you still experience this? I don't notice anything abnormal with trunk and Blink, and we had Timo's comment about it perhaps being a bug in SFLphone. |