similar to: Understanding Codecs

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>

I receive an INVITE/SDP containing: m=audio 11310 RTP/AVP 3 0 101 which I interpret as gsm, ulaw, rfc2833. and I reply with an OK/SDP containing: m=audio 15884 RTP/AVP 0 3 101 which I interpret as ulaw, gsm, rfc2833. How can I tell which codec was actually used for the call? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca

Dear list, I need to convert this data.frame > names(codesM) [1] "key" "AMR.pa1.M" "AMR.pa2.M" "AMR.pa3.M" "AMR.pa4.M" [6] "AMR.pa5.M" "AMR.pa6.M" "AMR.pa7.M" "AMR.pa8.M" "AMR.pa9.M" [11] "AMR.pa10.M" "AMR.ta1.M" "AMR.ta2.M" "AMR.ta3.M"

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Hi all, supposing to have speex working at 3.95, when VAD/CNG is employed, which is the size of those "silence" samples respect to voice samples? I am wondering if it could be possible to lower even the transmission of VAD/CNG samples by specifying "outbound" in the communication channel only 1 bit instead of sending the whole VAD/CNG packet. I am doing the same for AMR

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Hi All, First of all, I'm sorry if my question is offtopic on this list. In such case please ignore this post and/or contact me directly. I'm asking my questions there because I feel you had similar problem before starting developing Speex. My story: my friend developed 3gpp content creator and he would distribute it in binary form. But there is problem with AMR licensing (the terms

Hello, I'm trying to upgrade from 5-stable to 6-stable; after rebooting to the new kernel, the amr(4) module (amr.ko) refuses to load with this message: link_elf: symbol cam_simq_alloc undefined However, a quick nm(1) on /boot/kernel/cam.ko does show: ... 0000f0ac T cam_sim_set_path 0000efc4 T cam_simq_alloc 0000efd4 T cam_simq_free ... and cam.ko is already loaded before amr.ko.

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan->nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta

In order to develop a Voip application, today i should make it robust to bit-errors over wireless networks. This is an actually lack of Speex, infact what i've understood is that if a packet arrives corrupted, i must pass NULL to the decoder in order to let it know. My target is to use UDP (with checksum field disabled) and exploit also corrupted packets giving them to the AMR codec. Otherwise

Hi, R-users, data is a matrix like this AMR BS GE HR MO UK SP500 1974 -0.3505 -0.1154 -0.4246 -0.2107 -0.0758 0.2331 -0.2647 1975 0.7083 0.2472 0.3719 0.2227 0.0213 0.3569 0.3720 1976 0.7329 0.3665 0.2550 0.5815 0.1276 0.0781 0.2384 1977 -0.2034 -0.4271 -0.0490 -0.0938 0.0712 -0.2721 -0.0718 1978 0.1663 -0.0452 -0.0573 0.2751 0.1372 -0.1346

Hi I am using first time speex library, and this is my first problem. I need to decode AMR-NB packet to PCM. I read all manual instruction and I wrote these simple lines of code. bool CMMediaObj::AmrInit() { speex_bits_init(&bits); destate = speex_decoder_init(&speex_nb_mode); int tmp=1; speex_decoder_ctl(destate, SPEEX_SET_ENH,

Hello List; I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I am getting the following debug, can someone advise me about the solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE sip:22021782 at Asterisk_IP_Address:5060 SIP/2.0?Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1?From: "1828444" <sip:1828444 at

Hi all, Thanks to the previous replies that helped me with this before, but I got side-tracked in the middle of trying to figure this out, so apologies for posting the same issue. I use a Nokia e71, with an asterisk server and am having an issue dialing certain numbers. When I attempt to dial a local number, like xxx-xxx-xxxx, I cannot connect. What shows in the asterisk debug is the