Re: [asterisk-users] Delay on speak with Asterisk
Am 04.12.2019 um 11:14 schrieb Antony Stone: > Hm, I was judging based on what you posted previously: > > Our Codec Capability: (alaw) > Their Codec Capability: (ulaw|gsm|alaw|amr) > Joint Codec Capability: (alaw) > > which suggested to me that if you offered GSM, that could be agreed with the > other side. I tried again right now: disallow=all allow=alaw allow=gsm If I call my phone I can see, alaw is used. If I allow just gsm I get the error: [Dec 4 11:23:17] NOTICE[14060][C-012e]: chan_sip.c:10798 process_sdp: No compatible codecs, not accepting this offer! So, back to alaw... :( > Ah, but SIP is not RTP :) OK, I forgot it... I privilege RTP, too... ;) Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
On Wednesday 04 December 2019 at 11:00:23, Luca Bertoncello wrote: > Am 04.12.2019 um 10:53 schrieb Antony Stone: > > Hi Antony! > > > 1. Try using codec GSM (which is pretty good quality but lower bandwidth > > than alaw, which is currently the only one you are offering). > > gsm seems to be unsupported from Deutsche Telekom... > Already tried, it does not work... :( Hm, I was judging based on what you posted previously: Our Codec Capability: (alaw) Their Codec Capability: (ulaw|gsm|alaw|amr) Joint Codec Capability: (alaw) which suggested to me that if you offered GSM, that could be agreed with the other side. > > 2. What is the bandwidth (upstream is more important than downstream) of > > your Internet connection? > > Down 50Mbps > Up 10Mbps Well, that should certainly be plenty for a single VoIP channel (which I usually estimate as 100kpbs each way for ulaw or alaw). > On my Router (Debian 9) I configured a traffic shaper that privileges > the SIP-Packets. Ah, but SIP is not RTP :) SIP is used to set up the call, tell the other end what number you want to dial, tell you that the phone needs to ring, etc. It's not the audio part of the call once it's set up. RTP is the audio part of the call, and that's what you're saying is not so good now you've disabled the jitter buffer. RTP is UDP packets normally sent on any port between 10,000 and 20,000, so you need to ensure that your router allows that through with as low latency (and very importantly, consistent latency, since inconsistent latency = jitter) as possible. Prioritising SIP is hardly ever needed - who cares about a few tenths of a second setting up or responding to a call? What needs prioritising, and QoS if you can do it, is RTP. Antony. -- Police have found a cartoonist dead in his house. They say that details are currently sketchy. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
Am 04.12.2019 um 10:53 schrieb Antony Stone: Hi Antony! > 1. Try using codec GSM (which is pretty good quality but lower bandwidth than > alaw, which is currently the only one you are offering). gsm seems to be unsupported from Deutsche Telekom... Already tried, it does not work... :( > 2. What is the bandwidth (upstream is more important than downstream) of your > Internet connection? Down 50Mbps Up 10Mbps On my Router (Debian 9) I configured a traffic shaper that privileges the SIP-Packets. Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
On Wednesday 04 December 2019 at 07:37:51, Luca Bertoncello wrote: > Am 03.12.2019 um 19:28 schrieb Luca Bertoncello: > > Hi again > > > This delay happens on every peer, Deutsche Telekom and Messagenet, so I > > think the problem is NOT by the Provider, but in my configuration... > > Maybe I got the solution... > I see, that I had the jitter buffer active. As I deactivated it, I have > no delay anymore. > Unfortunately is the audio quality now a little bad than with the jitter > buffer... > > Any suggestion how can I improve the audio quality without add the delay? 1. Try using codec GSM (which is pretty good quality but lower bandwidth than alaw, which is currently the only one you are offering). 2. What is the bandwidth (upstream is more important than downstream) of your Internet connection? Antony. -- Why is "dylexia" so difficult to spell, and why can I never remember "aphasia" when I want to? Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
Am 03.12.2019 um 19:28 schrieb Luca Bertoncello: Hi again > This delay happens on every peer, Deutsche Telekom and Messagenet, so I > think the problem is NOT by the Provider, but in my configuration... Maybe I got the solution... I see, that I had the jitter buffer active. As I deactivated it, I have no delay anymore. Unfortunately is the audio quality now a little bad than with the jitter buffer... Any suggestion how can I improve the audio quality without add the delay? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
Am 03.12.2019 um 19:57 schrieb Antony Stone: Hi Antony, thank you for your answer. > I would firstly look at whether your Asterisk box is doing transcoding - > converting from oe codec (supported by your phones) and another codec > (supported by the provider) because no codec can be found in common between > the two. > > Secondly I would put a full packet sniffer (by which I mean collect all the > RTP > data as well as SIP) on each of your interfaces (internal and external) to > see > whether the delay really is happening inside your Asterisk server - if you > see > RTP data on your internal interface, then appearing 1-1.5 seconds later on > the > external interface, and vice versa, then you know the delay is inside your > system. I'm really not an expert on Asterisk... Could you please say me HOW can I check the codecs? I tried to get the information of the channel: bpi*CLI> sip show channel p65551t1575398506m6025c4749452s2 * SIP Call Curr. trans. direction: Incoming Call-ID:p65551t1575398506m6025c4749452s2 Owner channel ID: SIP/pbxanika-021e Our Codec Capability: (alaw) Non-Codec Capability (DTMF): 1 Their Codec Capability: (ulaw|gsm|alaw|amr) Joint Codec Capability: (alaw) Format: (alaw) T.38 supportNo Video support No MaxCallBR: 384 kbps Theoretical Address:217.x.x.x:5060 Received Address: 217.x.x.x:5060 SIP Transfer mode: open Force rport:Auto (No) Audio IP: 217.y.y.y (local) Our Tag:as45e11359 Their Tag: h7g4Esbg_p65551t1575398506m6025c4749452s1_206873930-910452977 SIP User agent: Username: 550293777072-0001 Peername: pbxanika Original uri: sip:sgc_c@217.x.y.z Caller-ID: +49177 Need Destroy: No Last Message: Rx: ACK Promiscuous Redir: No Route: DTMF Mode: rfc2833 SIP Options:timer Session-Timer: Inactive Transport: UDP Media: RTP Maybe it helps to find the problem? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay on speak with Asterisk
On Tuesday 03 December 2019 at 19:28:19, Luca Bertoncello wrote: > Hi list! > > I'm using Asterisk 13.14.1 from Debian 9 repositories. > The provider is Deutsche Telekom und Messagenet (just for receive). > > I can call and receive calls, but I have a little problem: there is a > "delay" of about 1-1,5 seconds between the time the voice is sent and > the time when the voice is received, so that it happens very often that > the peer does not get my voice and try to repeat the question, then it > get my voice, and breaks the question, and so on... > > This delay happens on every peer, Deutsche Telekom and Messagenet, so I > think the problem is NOT by the Provider, but in my configuration... > > Can someone suggest me where can I search the problem? I would firstly look at whether your Asterisk box is doing transcoding - converting from oe codec (supported by your phones) and another codec (supported by the provider) because no codec can be found in common between the two. Secondly I would put a full packet sniffer (by which I mean collect all the RTP data as well as SIP) on each of your interfaces (internal and external) to see whether the delay really is happening inside your Asterisk server - if you see RTP data on your internal interface, then appearing 1-1.5 seconds later on the external interface, and vice versa, then you know the delay is inside your system. No point in making such an assumption only to find out it's someone else doing the transcoding and introducing the delay there :) Hope that helps, Antony. -- Tinned food was developed for the British Navy in 1813. The tin opener was not invented until 1858. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
