Hello all! For several months now we?ve been experiencing a really strange problem with sound which best can be explained as choppy/stuttery, and with a touch of echo on top. Basically, parts of a conversation might be choppy, but often combined with some echo as well. The sound problem can only be heard by us, not by the other party. But this is the strange part: The problem only occurs periodically, and only towards one single SIP provider. All our equipment is SIP based (we use Cisco 79XX equipment with firmware 7.5 and 8.x), and we have no PSTN lines. Since this has been lasting for months, I don?t know the exact time the problem occurred at first. I recall everything as fine when I ran Asterisk 1.0.7, but through 1.0.9, 1.2.2 and 1.2.5 (HEAD) this problem persists. However, I have a hard time believing this is caused by a bug. Why? Well, as said, we have no problems with other providers. In addition, we have Cisco phones that communicate with Asterisk at the office (traffic is switched between two local subnets) and also over the internet, and all calls made between these phones are completed without any trouble. So I tried doing a clean install on a fresh CentOS last week, but with no luck ? the problem is still there. I also checked if there were load or IRQ issues, which there is not.The box runs APF firewall and has ports 5060-5070, as well as 8000-10000 (for rtp) open tcp/udp. From the phone information menu I can see that frame size is most likely set to 20 ms, while the provider uses 20 or 40 ms (not sure). I should add that this provider is among the larger ones in Norway (IP24). Other users have reported Asterisk to work just fine with it in the Provider?s local forum, but that is certainly not my experience. So my question is, is there anything I might have overlooked, or is there a particular way I can debug this issue?>From sip.conf:[general] port = 5060 bindaddr = 0.0.0.0 context = default disallow = all ;allow = ulaw allow = alaw ;allow = g729 allow=gsm maxexpirey = 3600 defaultexpirey = 160 externip = xx.xx.xx.xx // Asterisk?s static IP ;externrefresh = 60 notifymimetype=application/simple-message-summary dtmfmode= rfc2833 pedantic=no register => 11223344:password:11223344@sip.provider.com:5060/11223344 //phonenumber:password:phonenumber@sip-proxy:5060/extension [11223344] context= full-access type=friend secret=apassword fromuser=11223344 username=11223344 host=sip.provider.com fromdomain=sip.provider.com restrictid=yes canreinvite=no insecure=very deny=0.0.0.0/0.0.0.0 permit=123.123.123.0/255.255.255.0 (provider?s subnet) nat=no Thanks for all help! Regards, Bjorn -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.2.6/287 - Release Date: 21.03.2006 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060322/25977bf0/attachment.htm
On Mar 22, 2006, at 5:31 AM, Bjorn O wrote:> Hello all! > ? > For several months now we?ve been experiencing a really strange > problem with sound which best can be explained as choppy/stuttery, and > with a touch of echo on top. Basically, parts of a conversation might > be choppy, but often combined with some echo as well. The sound > problem can only be heard by us, not by the other party. But this is > the strange part: > ? > The problem only occurs periodically, and only towards one single SIP > provider. All our equipment is SIP based (we use Cisco 79XX equipment > with firmware 7.5 and 8.x), and we have no PSTN lines. Since this has > been lasting for months, I don?t know the exact time the problem > occurred at first. I recall everything as fine when I ran Asterisk > 1.0.7, but through 1.0.9, 1.2.2 and 1.2.5 (HEAD) this problem > persists. However, I have a hard time believing this is caused by a > bug. Why? Well, as said, we have no problems with other providers. In > addition, we have Cisco phones that communicate with Asterisk at the > office (traffic is switched between two local subnets) and also over > the internet, and all calls made between these phones are completed > without any trouble. > ? > So I tried doing a clean install on a fresh CentOS last week, but with > no luck ? the problem is still there. I also checked if there were > load or IRQ issues, which there is not.The box runs APF firewall and > has ports 5060-5070, as well as 8000-10000 (for rtp) open tcp/udp. > From the phone information menu I can see that frame size is most > likely set to 20 ms, while the provider uses 20 or 40 ms (not sure). > ? > I should add that this provider is among the larger ones in Norway > (IP24). Other users have reported Asterisk to work just fine with it > in the Provider?s local forum, but that is certainly not my > experience. So my question is, is there anything I might have > overlooked, or is there a particular way I can debug this issue? > ?Sounds like a jitter issue, based on your description. When this happens have you looked at a traceroute to the SIP call terminator? Perhaps there is a flakey router in your path? Marty -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 3460 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060322/f62a3d99/attachment.bin
When you hear echo its actually being caused by the other phone. check your zapata.conf file and see if echotraining=yes. Echo can be caused by lots of things so this is just a starting point. For several months now we?ve been experiencing a really strange problem with sound which best can be explained as choppy/stuttery, and with a touch of echo on top. Basically, parts of a conversation might be choppy, but often combined with some echo as well. The sound problem can only be heard by us, not by the other party. But this is the strange part: The problem only occurs periodically, and only towards one single SIP provider. All our equipment is SIP based (we use Cisco 79XX equipment with firmware 7.5 and 8.x), and we have no PSTN lines. Since this has been lasting for months, I don?t know the exact time the problem occurred at first. I recall everything as fine when I ran Asterisk 1.0.7, but through 1.0.9, 1.2.2 and 1.2.5 (HEAD) this problem persists. However, I have a hard time believing this is caused by a bug. Why? Well, as said, we have no problems with other providers. In addition, we have Cisco phones that communicate with Asterisk at the office (traffic is switched between two local subnets) and also over the internet, and all calls made between these phones are completed without any trouble.
But is Zapata.conf even in the picture when all calls go through a SIP-provider, since the calls are not being routed through a ZAP channel? Regards, Bjorn -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av steve Sendt: 22. mars 2006 23:11 Til: asterisk-users@lists.digium.com Emne: [Asterisk-Users] re: Sound issues on SIP-SIP calls When you hear echo its actually being caused by the other phone. check your zapata.conf file and see if echotraining=yes. Echo can be caused by lots of things so this is just a starting point. For several months now we?ve been experiencing a really strange problem with sound which best can be explained as choppy/stuttery, and with a touch of echo on top. Basically, parts of a conversation might be choppy, but often combined with some echo as well. The sound problem can only be heard by us, not by the other party. But this is the strange part: The problem only occurs periodically, and only towards one single SIP provider. All our equipment is SIP based (we use Cisco 79XX equipment with firmware 7.5 and 8.x), and we have no PSTN lines. Since this has been lasting for months, I don?t know the exact time the problem occurred at first. I recall everything as fine when I ran Asterisk 1.0.7, but through 1.0.9, 1.2.2 and 1.2.5 (HEAD) this problem persists. However, I have a hard time believing this is caused by a bug. Why? Well, as said, we have no problems with other providers. In addition, we have Cisco phones that communicate with Asterisk at the office (traffic is switched between two local subnets) and also over the internet, and all calls made between these phones are completed without any trouble. _______________________________________________ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.2.6/288 - Release Date: 22.03.2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.385 / Virus Database: 268.2.6/288 - Release Date: 22.03.2006