Re: [asterisk-users] Re: IAX2 goes one way audio when lag gets bad
if anyone has one-way audio issues with iax over jittery connection, please look at bug report, what I created yesterday and report your experiences, I think this is one of the most serious bug, that must be identified and resolved before 1.4 will be released, thanks http://bugs.digium.com/view.php?id=8325 PJ Benjamin Jacob wrote: Martin Joseph wrote: On 2006-10-25 08:14:43 -0700, Noah Miller [EMAIL PROTECTED] said: Hi Matt - I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. On the customer's end I have the following config in iax.conf: trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 If you're using Asterisk 1.2.x, dropcount, jittershrinkrate and minexcesbuffer don't do anything. They are ignored by 1.2.x unless you specify that you want to use the old 1.0.x jitterbuffer. Instead you might try the parameters maxjitterbuffer, resyncthreshold, and maxjitterinterps. For more, you can check out the sample iax.conf. I believe, also, that you are correct in setting trunk=no. I know in the 1.0.x jitterbuffer, trunk was not fully supported. I think this is still the case with the 1.2.x jitterbuffer. If the audio is dropping out completely, then I suspect the whole jitter buffer thing is a red herring (waste of time). Perhaps it's a nat issue? What kind of router if any is involved? I am reaching here... Also, please do tell us which version of asterisk you are running... Marty seeing this thread a lil too late, i guess. So, am sorry if I am repeating things. When I was setting up my iax2 configs, I too had one way audio initialy. Tried the softphone on two machines(which incidentaly had asterisk running on them as well), to no avail. When I looked at the tcpdump on my asterisk server, I could see no rtp coming in from the two said machines. So, I shifted the softphone to another machine, this time on a windows machine, n voila! it worked like a charm. So, I hope you did have a look at the tcpdump to check on the rtp flow. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: IAX2 goes one way audio when lag gets bad
On 2006-10-24 13:04:02 -0700, Matt [EMAIL PROTECTED] said: Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw notransfer=yes trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 I have tried with jitterbuffer=no, and then rather then one-way-audio I get high packet loss until the connection settles back down.Any ideas on other things I can try? Implement QoS that prevents the upstream bandwidth from the customers site from being completely hammered... Just a thought, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: IAX2 goes one way audio when lag gets bad
Marty, Thanks for the suggestion... unfortunately it is not a case of the bandwidth being hammered. The only things on this connection is the voice.My thought is there is something wrong, possibly, with the cable provider's node. Still.. Asterisk shouldn't just barf with one-way-audio. On 10/25/06, Martin Joseph [EMAIL PROTECTED] wrote: On 2006-10-24 13:04:02 -0700, Matt [EMAIL PROTECTED] said: Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw notransfer=yes trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 I have tried with jitterbuffer=no, and then rather then one-way-audio I get high packet loss until the connection settles back down.Any ideas on other things I can try? Implement QoS that prevents the upstream bandwidth from the customers site from being completely hammered... Just a thought, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: IAX2 goes one way audio when lag gets bad
Pavel, What version of asterisk are you connecting back to? Is it also 1.4. branch? On 10/25/06, Pavel Jezek [EMAIL PROTECTED] wrote: I have same problem, but only with 1.4 branch and when some bigger jitter occur (1.2 is working fine, even in case with big jitter), I dump packets with tcpdump and see, that asterisk stops sending packets in one direction... Maybe good reason to open bug report for this, because QoS settings ins not always possible (e.g. my case with CDMA connection) PJ Martin Joseph wrote: On 2006-10-24 13:04:02 -0700, Matt [EMAIL PROTECTED] said: Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw notransfer=yes trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 I have tried with jitterbuffer=no, and then rather then one-way-audio I get high packet loss until the connection settles back down.Any ideas on other things I can try? Implement QoS that prevents the upstream bandwidth from the customers site from being completely hammered... Just a thought, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: IAX2 goes one way audio when lag gets bad
On 2006-10-25 08:14:43 -0700, Noah Miller [EMAIL PROTECTED] said: Hi Matt - I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. On the customer's end I have the following config in iax.conf: trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 If you're using Asterisk 1.2.x, dropcount, jittershrinkrate and minexcesbuffer don't do anything. They are ignored by 1.2.x unless you specify that you want to use the old 1.0.x jitterbuffer. Instead you might try the parameters maxjitterbuffer, resyncthreshold, and maxjitterinterps. For more, you can check out the sample iax.conf. I believe, also, that you are correct in setting trunk=no. I know in the 1.0.x jitterbuffer, trunk was not fully supported. I think this is still the case with the 1.2.x jitterbuffer. If the audio is dropping out completely, then I suspect the whole jitter buffer thing is a red herring (waste of time). Perhaps it's a nat issue? What kind of router if any is involved? I am reaching here... Also, please do tell us which version of asterisk you are running... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: IAX2 goes one way audio when lag gets bad
If the audio is dropping out completely, then I suspect the whole jitter buffer thing is a red herring (waste of time). Perhaps it's a nat issue? What kind of router if any is involved? I am reaching here... Also, please do tell us which version of asterisk you are running... I apologize.. I thought I told already. I am running 1.2.6 and have tried 1.2.12. At any rate, I believe it is actually the cable modem connection dropping, and someone from Comcast is coming to look at it tomorrow. My question is.. why is the jitterbuffer just dieing? I understand there may not be audio if the connection dropped for like 4 or 5 seconds, but shouldn't it pick back up? Using pingplotter I've determined when they are losing audio, I also get a red 100% packet loss from their node lasts about 5 seconds usually, and then the jitterbuffer is dead. Is this just too much for the jitterbuffer to handle? Can't it get back on track at least? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: IAX2 goes one way audio when lag gets bad
Martin Joseph wrote: On 2006-10-25 08:14:43 -0700, Noah Miller [EMAIL PROTECTED] said: Hi Matt - I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. On the customer's end I have the following config in iax.conf: trunk=no (I have also tried trunk=yes and nothing for trunk=) jitterbuffer=yes forcejitterbuffer=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 If you're using Asterisk 1.2.x, dropcount, jittershrinkrate and minexcesbuffer don't do anything. They are ignored by 1.2.x unless you specify that you want to use the old 1.0.x jitterbuffer. Instead you might try the parameters maxjitterbuffer, resyncthreshold, and maxjitterinterps. For more, you can check out the sample iax.conf. I believe, also, that you are correct in setting trunk=no. I know in the 1.0.x jitterbuffer, trunk was not fully supported. I think this is still the case with the 1.2.x jitterbuffer. If the audio is dropping out completely, then I suspect the whole jitter buffer thing is a red herring (waste of time). Perhaps it's a nat issue? What kind of router if any is involved? I am reaching here... Also, please do tell us which version of asterisk you are running... Marty seeing this thread a lil too late, i guess. So, am sorry if I am repeating things. When I was setting up my iax2 configs, I too had one way audio initialy. Tried the softphone on two machines(which incidentaly had asterisk running on them as well), to no avail. When I looked at the tcpdump on my asterisk server, I could see no rtp coming in from the two said machines. So, I shifted the softphone to another machine, this time on a windows machine, n voila! it worked like a charm. So, I hope you did have a look at the tcpdump to check on the rtp flow. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
