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 Bertoncello (lucabert at lucabert.de)
On Saturday 13 June 2020 at 13:36:00, Luca Bertoncello wrote:> 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).What does that mean? You're making a mobile phone call over the GSM network to your Asterisk server, or you're using a soft phone application on your smartphone, which is registered by SIP to your Asterisk server? Also, where did you make the call *to* ?> The quality was top... > Maybe is the problem in a codec used from our phones at homes?Didn't we already discuss this last year? http://lists.digium.com/pipermail/asterisk-users/2019-December/294446.html> Could someone suggest me how to check the codec used by my mobile phone > and the codec used by the phones at home?Look at the verbose log file and search for "transcoding". Also, do a SIP packet trace at the start of the call and see which codecs are announced by each side and then what gets agreed on (I don't think this gets logged by Asterisk, so you need to look at the SIP negotiation itself). Antony. -- I'm not impossible, just highly implausible. Please reply to the list; please *don't* CC me.
> Am 13.06.2020 um 13:36 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > 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 Bertoncello > (lucabert at lucabert.de)Try "sip show peer <peername>" for a phone. Then "sip show channels" during an existing call. And "sip show channel <Call-ID>" for more info. Michael http://www.mksolutions.info
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 Record Off feature : automon Subscr.Cont. : <Not set> Language : de Tonezone : <Not set> AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : 1 Pickupgroup : 1 Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "0049177xxxxxxx" <> MaxCallBR : 384 kbps Expire : -1 Insecure : no Force rport : Yes Symmetric RTP: Yes ACL : No DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes Path support : No Path : N/A TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : (null) Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk) Auto-Framing : No Status : UNKNOWN Useragent : Reg. Contact : Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Refuse Sess-Refresh : uac Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No VoIP-phone (Thomson ST2022): bpi*CLI> sip show peer 0049351xxxxxxx * Name : 0049351xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon Record Off feature : automon Subscr.Cont. : <Not set> Language : de Tonezone : <Not set> AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : 1 Pickupgroup : 1 Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "0049351xxxxxxx" <> MaxCallBR : 384 kbps Expire : 3111 Insecure : no Force rport : Yes Symmetric RTP: Yes ACL : Yes DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : Yes Path support : No Path : N/A TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 192.168.200.10:25572 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 0049351xxxxxxx SIP Options : (none) Codecs : (alaw|ulaw|ilbc|g729|g723|gsm) Auto-Framing : No Status : OK (17 ms) Useragent : THOMSON ST2022 hw2 fw3.56 00-26-44-31-10-23 Reg. Contact : sip:0049351xxxxxxx at 192.168.200.10:25572;user=phone Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Refuse Sess-Refresh : uac Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No> Then "sip show channels" during an existing call.Call from normal phone: bpi*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 192.168.200.10 0049351xxxxxxx 9eff88f7-c0a801 (alaw) No Rx: ACK 0049351xxxxxxx 217.0.27.53 03501xxxxxxx 453efbcb7a04f33 (alaw) No Tx: ACK pbxluca 2 active SIP dialogs Call from mobile phone (via VoIP registered in Asterisk): bpi*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 192.168.10.12 0049177xxxxxxx 11b86bd612b71ae (alaw) No Rx: INVITE 0049177xxxxxxx 217.0.27.53 00493501xxxxxxx 5647efe41d746b4 (alaw) No Tx: INVITE pbxluca 2 active SIP dialogs> And "sip show channel <Call-ID>" for more info.Call from normal phone: bpi*CLI> sip show channel 9eff88f7-c0a80101-0-22c911 at 192.168.200.10 * SIP Call Curr. trans. direction: Incoming Call-ID: 9eff88f7-c0a80101-0-22c911 at 192.168.200.10 Owner channel ID: SIP/0049351xxxxxxx-000000a7 Our Codec Capability: (alaw|ulaw|ilbc|g729|g723|gsm) Non-Codec Capability (DTMF): 1 Their Codec Capability: (ulaw|g723|alaw|g729) Joint Codec Capability: (alaw|ulaw|g729|g723) Format: (alaw) T.38 support No Video support No MaxCallBR: 384 kbps Theoretical Address: 192.168.200.10:25572 Received Address: 192.168.200.10:25572 SIP Transfer mode: open Force rport: Yes Audio IP: 192.168.200.1 (local) Our Tag: as12e44b1b Their Tag: c0a80101-d3c8cef7 SIP User agent: THOMSON ST2022 hw2 fw3.56 00-26-44-31-10-23 Username: 0049351xxxxxxx Peername: 0049351xxxxxxx Original uri: sip:0049351xxxxxxx at 192.168.200.10:25572 Caller-ID: 0049351xxxxxxx Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: <sip:0049351xxxxxxx at 192.168.200.10:25572;user=phone> DTMF Mode: rfc2833 SIP Options: replaces replace timer Session-Timer: Inactive Transport: UDP Media: RTP bpi*CLI> sip show channel 453efbcb7a04f33e1e0de7ef461f9b38 at tel.t-online.de * SIP Call Curr. trans. direction: Outgoing Call-ID: 453efbcb7a04f33e1e0de7ef461f9b38 at tel.t-online.de Owner channel ID: SIP/pbxluca-000000a8 Our Codec Capability: (alaw|ulaw) Non-Codec Capability (DTMF): 1 Their Codec Capability: (alaw) Joint Codec Capability: (alaw) Format: (alaw) T.38 support Yes Video support No MaxCallBR: 384 kbps Theoretical Address: 217.0.27.xx:5060 Received Address: 217.0.27.xx:5060 SIP Transfer mode: open Force rport: Yes Audio IP: 91.49.50.x (local) Our Tag: as29bbbfb6 Their Tag: h7g4Esbg_p65551t1592060241m195254c7230720s1_1763914935-920913141 SIP User agent: Username: 03501xxxxxxx Peername: pbxluca Original uri: sip:sgc_c at 217.0.27.xx Need Destroy: No Last Message: Tx: ACK Promiscuous Redir: No Route: <sip:217.0.27.xx;transport=udp;lr> DTMF Mode: rfc2833 SIP Options: (none) Session-Timer: Inactive Transport: UDP Media: RTP Call from mobile phone (via VoIP registered in Asterisk): bpi*CLI> sip show channel 11b86bd612b71ae0f06c62d53ecf08c6 at 192.168.10.12 * SIP Call Curr. trans. direction: Incoming Call-ID: 11b86bd612b71ae0f06c62d53ecf08c6 at 192.168.10.12 Owner channel ID: SIP/0049177xxxxxxx-000000a9 Our Codec Capability: (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk) Non-Codec Capability (DTMF): 1 Their Codec Capability: (ulaw|gsm|alaw|amr) Joint Codec Capability: (alaw|ulaw|gsm|amr) Format: (alaw) T.38 support No Video support No MaxCallBR: 384 kbps Theoretical Address: 192.168.10.12:37210 Received Address: 192.168.10.12:37210 SIP Transfer mode: open Force rport: Yes Audio IP: 192.168.10.1 (local) Our Tag: as339b5367 Their Tag: 1910565801 SIP User agent: Peername: 0049177xxxxxxx Original uri: sip:0049177xxxxxxx at 192.168.10.12:37210 Caller-ID: 0049177xxxxxxx Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: <sip:0049177xxxxxxx at 192.168.10.12:37210;transport=udp> DTMF Mode: rfc2833 SIP Options: (none) Session-Timer: Inactive Transport: UDP Media: RTP bpi*CLI> sip show channel 5647efe41d746b4d67ad5c576b67beba at tel.t-online.de * SIP Call Curr. trans. direction: Outgoing Call-ID: 5647efe41d746b4d67ad5c576b67beba at tel.t-online.de Owner channel ID: SIP/pbxluca-000000aa Our Codec Capability: (alaw|ulaw) Non-Codec Capability (DTMF): 1 Their Codec Capability: (alaw) Joint Codec Capability: (alaw) Format: (alaw) T.38 support Yes Video support No MaxCallBR: 384 kbps Theoretical Address: 217.0.27.xx:5060 Received Address: 217.0.27.xx:5060 SIP Transfer mode: open Force rport: Yes Audio IP: 91.49.50.xx (local) Our Tag: as148b6300 Their Tag: h7g4Esbg_p65551t1592060364m136229c7238384s1_1886856096-203650581 SIP User agent: Username: 00493501xxxxxxx Peername: pbxluca Original uri: sip:sgc_c at 217.0.27.xx Need Destroy: No Last Message: Tx: ACK Promiscuous Redir: No Route: <sip:217.0.27.xx;transport=udp;lr> DTMF Mode: rfc2833 SIP Options: (none) Session-Timer: Inactive Transport: UDP Media: RTP So, I'd say, the codecs are the same... Do you see something strange that I should check/change? Thank you very very much for your help! Luca Bertoncello (lucabert at lucabert.de)