Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?
On 28/10/2013 4:12 PM, Mark Wiater wrote: On 10/28/2013 3:59 PM, Ron Wheeler said: I am reaching the same level of frustration. I have tried to find the source of the problems. We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk - No analogue. I don't have any problems with IAX, but I hear some do. I have now switched to SIP and will check the quality in the morning. We have a very lightly loaded 60 Mbs cable link to the Internet that tests pretty close to that most of the time. Bandwidth is less important than the overall quality of the internet link, latency and jitter. Either way, there is no QoS on the internet, all bets are off. The codec can matter too. What are you using? G711 I have not found any good tools to track down the causes of poor voice quality. In my case, I have good incoming quality and terrible quality going out. Oh, is your cable connection assymetric? Upload smaller than download? If so, that correlates to terrible audio, right? Just ran a test 50 Mbps download 10Mbps upload. Should be enough I hope. That is, I can hear people perfectly well but they complain that my voice drops out and is garbled regardless of who places the call. As a result, I use Skype for all of my calls and if someone calls me, I call them back on Skype if they have any problems. I don't understand why Skype works so well and Asterisk works so poorly on the same environment. Googling Asterisk poor audio quality return several hundred thousand references I'd not shoot asterisk yet. I'd focus on the internet connection and it's components (cable modem) first. Good idea. I am sure that you are right but what to test and how are not clear. I use asterisk all over the place. Mostly connected to PRI's and Carrier provided SIP trunks, with internet SIP trunks as backup. I get complaints on the Internet based SIP trunks sometimes, never on other other two. I'd ask most of these questions of the OP too. Overall telephony design matters. -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?
On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote: All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure what option would be the best Put analog lines in the conference room to avoid the dropouts - leave the sip lines in place for day to day use Hire a consultant Ditch the system and buy a pre-packaged system - RingCentral or some such. There are no local asterisk professionals who can help, and we are a little leery of opening up our system to outside consultants. Anyone else face the above, and finally abandoned Asterisk for a commercial system? We have 167 users. I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference rooms. Suggestions welcome. A general rule of thump after several years with voip Voip turns out to be the canary in the coal-mine of a network. The smallest change or problem will manifest itself as a voip issue no matter what. Now to some practical advice Voip was designed for LAN's, The moment voip packets leave your lan and go into a WAN of any sort, it could be the source of frustration for many reasons. 1) Lots of routers/modems are not build to handle intense voip traffic. voip generates lots of small in size UPD packages. In most of the cases the routers/modems bridging your lan with the wan have no problem handling them BUT what i have found is that once you get over a threshold of traffic its possible the routers/modem can not cope with it, mainly because the large number of packets they have to process. In most enterprise grade routers the specs give you 2 numbers for the size of data the router can handle. total throughput and pps (packets per second). Usually total throughput is calculated using a packet size of around 1500bytes and it takes the router the same resources to process a 1500 bytes package as it does a 90bytes packet of a g729 call, as it just looks at the headers and not the payload.So yes your router can handle 60Mbits (of 1500byte frames) which is about 5000 packers per second but for voip that translates to less than 4Mbits of data (5000 packets of 90 bytes) I think you can get the picture 2) Because of 1) its possible that your ISP has issues, especially if its handling lots of voip traffic while its equipment is not optimized for that. 3) QOS and queing in general Whatever you do with QOS to get a better priority/quality, the dirty secret is, you can only control what YOU send, not what you receive. And even that is true till your modem/router. Once the packet is gone you have no control of how it will be handle by all intermediates till it reaches its destination. You have no idea if qos is honored by ALL hops and what kind of queuing they apply (if they do) to that port/service/qos mark That beeing said, its possible that you *might* have much better luck with sip and sip rtp than with iax rtp if your isp and all its interconnects bother to offer qos for rtp. Now for receiving it can be even harder if your isp does not provide correct priority queuing for the rtp stream, as latencies can build fast especially on busy hours (which happen to be the same hours people use their phones the most...) where people download stuff,emails etc. ping.icmp and all the other networking monitoring tools/protocols could be an indicator BUT its most probable that they will be handled by the isp and its interconnects at the higher qos priority The only way to see how rtp traffic is handled is to run rtp traffic. The only way around this is a dedicated circut MPLS or similar between the points of interest (i.e offices), with specific SLA which usually means much much higher costs. Finally my 2 cents for troubleshouting. Check the network first ! Find what triggers the problem. Is it something that happens all time regardless of traffic ? is it periodic ? (when bw goes over X percent, or at a specific time of day ?) Try different qos settings/priority queuing on the router -- Stelios S. Koroneos Phone US : (+1) 347-783-5467 Greece : (+30) 211-800-7655 ext 101 Skype : skoroneos PGP Key fingerprint = DC66 109A 6C3A 2D65 BA52 806E 6122 DAF4 32E7 076A smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?
Hi there, Sounds like codec ptime mismatch...what codec are you using? If you are using g729 make sure that you and your provider is giving the same ptime. On 10/29/2013 11:55 AM, Stelios Koroneos wrote: On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote: All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure what option would be the best Put analog lines in the conference room to avoid the dropouts - leave the sip lines in place for day to day use Hire a consultant Ditch the system and buy a pre-packaged system - RingCentral or some such. There are no local asterisk professionals who can help, and we are a little leery of opening up our system to outside consultants. Anyone else face the above, and finally abandoned Asterisk for a commercial system? We have 167 users. I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference rooms. Suggestions welcome. A general rule of thump after several years with voip Voip turns out to be the canary in the coal-mine of a network. The smallest change or problem will manifest itself as a voip issue no matter what. Now to some practical advice Voip was designed for LAN's, The moment voip packets leave your lan and go into a WAN of any sort, it could be the source of frustration for many reasons. 1) Lots of routers/modems are not build to handle intense voip traffic. voip generates lots of small in size UPD packages. In most of the cases the routers/modems bridging your lan with the wan have no problem handling them BUT what i have found is that once you get over a threshold of traffic its possible the routers/modem can not cope with it, mainly because the large number of packets they have to process. In most enterprise grade routers the specs give you 2 numbers for the size of data the router can handle. total throughput and pps (packets per second). Usually total throughput is calculated using a packet size of around 1500bytes and it takes the router the same resources to process a 1500 bytes package as it does a 90bytes packet of a g729 call, as it just looks at the headers and not the payload.So yes your router can handle 60Mbits (of 1500byte frames) which is about 5000 packers per second but for voip that translates to less than 4Mbits of data (5000 packets of 90 bytes) I think you can get the picture 2) Because of 1) its possible that your ISP has issues, especially if its handling lots of voip traffic while its equipment is not optimized for that. 3) QOS and queing in general Whatever you do with QOS to get a better priority/quality, the dirty secret is, you can only control what YOU send, not what you receive. And even that is true till your modem/router. Once the packet is gone you have no control of how it will be handle by all intermediates till it reaches its destination. You have no idea if qos is honored by ALL hops and what kind of queuing they apply (if they do) to that port/service/qos mark That beeing said, its possible that you *might* have much better luck with sip and sip rtp than with iax rtp if your isp and all its interconnects bother to offer qos for rtp. Now for receiving it can be even harder if your isp does not provide correct priority queuing for the rtp stream, as latencies can build fast especially on busy hours (which happen to be the same hours people use their phones the most...) where people download stuff,emails etc. ping.icmp and all the other networking monitoring tools/protocols could be an indicator BUT its most probable that they will be handled by the isp and its interconnects at the higher qos priority The only way to see how rtp traffic is handled is to run rtp traffic. The only way around this is a dedicated circut MPLS or similar between the points of interest (i.e offices), with specific SLA which usually means much much higher costs. Finally my 2 cents for troubleshouting. Check the network first ! Find what triggers the problem. Is it something that happens all time regardless of traffic ? is it periodic ? (when bw goes over X percent, or at a specific time of day ?) Try different qos settings/priority queuing on the router -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To
Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?
Hi there, In other words you are maybe on 60ms and they are on 20ms or vice versa. Do a wireshark trace and see if the codecs and ptime agree on both sides otherwise you will get grabbled sounds. On 10/29/2013 02:49 PM, Daniel van den Berg wrote: Hi there, Sounds like codec ptime mismatch...what codec are you using? If you are using g729 make sure that you and your provider is giving the same ptime. On 10/29/2013 11:55 AM, Stelios Koroneos wrote: On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote: All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure what option would be the best Put analog lines in the conference room to avoid the dropouts - leave the sip lines in place for day to day use Hire a consultant Ditch the system and buy a pre-packaged system - RingCentral or some such. There are no local asterisk professionals who can help, and we are a little leery of opening up our system to outside consultants. Anyone else face the above, and finally abandoned Asterisk for a commercial system? We have 167 users. I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference rooms. Suggestions welcome. A general rule of thump after several years with voip Voip turns out to be the canary in the coal-mine of a network. The smallest change or problem will manifest itself as a voip issue no matter what. Now to some practical advice Voip was designed for LAN's, The moment voip packets leave your lan and go into a WAN of any sort, it could be the source of frustration for many reasons. 1) Lots of routers/modems are not build to handle intense voip traffic. voip generates lots of small in size UPD packages. In most of the cases the routers/modems bridging your lan with the wan have no problem handling them BUT what i have found is that once you get over a threshold of traffic its possible the routers/modem can not cope with it, mainly because the large number of packets they have to process. In most enterprise grade routers the specs give you 2 numbers for the size of data the router can handle. total throughput and pps (packets per second). Usually total throughput is calculated using a packet size of around 1500bytes and it takes the router the same resources to process a 1500 bytes package as it does a 90bytes packet of a g729 call, as it just looks at the headers and not the payload.So yes your router can handle 60Mbits (of 1500byte frames) which is about 5000 packers per second but for voip that translates to less than 4Mbits of data (5000 packets of 90 bytes) I think you can get the picture 2) Because of 1) its possible that your ISP has issues, especially if its handling lots of voip traffic while its equipment is not optimized for that. 3) QOS and queing in general Whatever you do with QOS to get a better priority/quality, the dirty secret is, you can only control what YOU send, not what you receive. And even that is true till your modem/router. Once the packet is gone you have no control of how it will be handle by all intermediates till it reaches its destination. You have no idea if qos is honored by ALL hops and what kind of queuing they apply (if they do) to that port/service/qos mark That beeing said, its possible that you *might* have much better luck with sip and sip rtp than with iax rtp if your isp and all its interconnects bother to offer qos for rtp. Now for receiving it can be even harder if your isp does not provide correct priority queuing for the rtp stream, as latencies can build fast especially on busy hours (which happen to be the same hours people use their phones the most...) where people download stuff,emails etc. ping.icmp and all the other networking monitoring tools/protocols could be an indicator BUT its most probable that they will be handled by the isp and its interconnects at the higher qos priority The only way to see how rtp traffic is handled is to run rtp traffic. The only way around this is a dedicated circut MPLS or similar between the points of interest (i.e offices), with specific SLA which usually means much much higher costs. Finally my 2 cents for troubleshouting. Check the network first ! Find what triggers the problem. Is it something that happens all time regardless of traffic ? is it periodic ? (when bw goes over X percent, or at a specific time of day ?) Try different qos settings/priority queuing on the router --
[asterisk-users] Loosing synch between party 1 party 2 voice in monitor recording
Hi We have come across a situation where we are loosing synch of party 1 party 2 voice in call recording. Here is the scenario Party 1 initiate a call to Party 2 using AMI commands When both calls are connected, we bridge these 2 calls. Then we start recording of this bridged call using AMI Monitor command. Monitor command is invoked on Party 1 only. Then we put party 2 on hold. We redirect party 2 to async AGI with AMI command. After few seconds, party 2 call is retrieved using AMI command. Party 1 Party 2 calls are again bridged. When call is terminated, monitor is stopped. Now in this scenario, party 1 party 2 voice has lost synch Call StateConnectedParty 2 on holdCall retrieved Terminated 0 sec25 sec 35 sec50 sec Party 1RecordingRecording Recording Recording stopped Party 2RecordingNot recording Recording Recording stopped Now in this case,I get recording of 50 sec. Where asparty 2 voice continue even after 25 sec even though party 2 was kept on hold. And party 2 side recording ends after 40 sec only. This way comminication sync is completely lost. Has some one come across such situation? Please help me to solve this issue. -- Thanks Regards, Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about how Asterisk works with RTP ports
Jonas Kellens wrote: Hello, short question : does Asterisk reserve RTP ports for every IP-phone that is being called ? It uses 2 ports per channel under normal circumstances, 1 for RTP and 1 for RTCP. If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ? Yes. I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this means at least 10 RTP ports are reserved for incoming audio, correct ??? Yes. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about how Asterisk works with RTP ports
Hello, short question : does Asterisk reserve RTP ports for every IP-phone that is being called ? If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ? I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this means at least 10 RTP ports are reserved for incoming audio, correct ??? Thanks. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about how Asterisk works with RTP ports
On 10/29/2013 05:14 PM, Joshua Colp wrote: Jonas Kellens wrote: Hello, short question : does Asterisk reserve RTP ports for every IP-phone that is being called ? It uses 2 ports per channel under normal circumstances, 1 for RTP and 1 for RTCP. If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ? Yes. I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this means at least 10 RTP ports are reserved for incoming audio, correct ??? Yes. So if I understand correct, you don't need to look at the amount of concurrent calls to calculate the RTP range in rtp.conf, you need to look at the amount of INVITES that are being send at one moment ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about how Asterisk works with RTP ports
Jonas Kellens wrote: So if I understand correct, you don't need to look at the amount of concurrent calls to calculate the RTP range in rtp.conf, you need to look at the amount of INVITES that are being send at one moment ? The number of concurrent channels in existence which are using RTP. While a channel may not be answered, it's still in existence. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No of parked calls limit
Is there a limit to the number of parked calls Asterisk can handle? Thanks, Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
