[Asterisk-Users] top-notch servers/OS/network, ulaw codec - sound still choppy
Hi, My SIP calls are sounding a little choppy. I've did my research but everything looks right on my end...what am I missing? Running RedHat ES 3.0 on dual AMD Opteron servers. My system is cololocated in downtown LA and is fed via a gigabit handoff from XO, ATT, Level 3 and Wiltel (I have a 100Mb didicated line). So I dont think its the Servers, its the network, Asterisk is working fine and all codecs look right...what could be the cause? **SNIP FROM SIP.CONF*** [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) ;bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records allow=ulaw ; Allow codecs in order of preference * ga0*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 64.201.99.2479092479878 2fd496bf330 00103/00105 ulaw ga0*CLI show version Asterisk 1.0.3 built by [EMAIL PROTECTED] on a x86_64 running Linux P.S. in my sip.conf file, it looks like I am only allowing the ulaw codec...could that cause a problem if I happen to need to call somebody that doesn't support ulaw? Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - sound still choppy
I see the sip user is an external ip. I would take a look at your QoS settings on your router. Make sure the voice traffic is getting the priority it deserves. Also, check for packet loss. On Thu, 20 Jan 2005 11:26:02 -0800, Gabriel Afana [EMAIL PROTECTED] wrote: Hi, My SIP calls are sounding a little choppy. I've did my research but everything looks right on my end...what am I missing? Running RedHat ES 3.0 on dual AMD Opteron servers. My system is cololocated in downtown LA and is fed via a gigabit handoff from XO, ATT, Level 3 and Wiltel (I have a 100Mb didicated line). So I dont think its the Servers, its the network, Asterisk is working fine and all codecs look right...what could be the cause? **SNIP FROM SIP.CONF*** [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) ;bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records allow=ulaw ; Allow codecs in order of preference * ga0*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 64.201.99.2479092479878 2fd496bf330 00103/00105 ulaw ga0*CLI show version Asterisk 1.0.3 built by [EMAIL PROTECTED] on a x86_64 running Linux P.S. in my sip.conf file, it looks like I am only allowing the ulaw codec...could that cause a problem if I happen to need to call somebody that doesn't support ulaw? Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - sound still choppy
Great, thanks for the info. This is a service provided from my colo, so I will have to give them a call and find out whats up with their router settings. As for packet loss, how do I check for that? Gabe - Original Message - From: Jon Radon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 20, 2005 12:32 PM Subject: Re: [Asterisk-Users] top-notch servers/OS/network,ulaw codec - sound still choppy I see the sip user is an external ip. I would take a look at your QoS settings on your router. Make sure the voice traffic is getting the priority it deserves. Also, check for packet loss. On Thu, 20 Jan 2005 11:26:02 -0800, Gabriel Afana [EMAIL PROTECTED] wrote: Hi, My SIP calls are sounding a little choppy. I've did my research but everything looks right on my end...what am I missing? Running RedHat ES 3.0 on dual AMD Opteron servers. My system is cololocated in downtown LA and is fed via a gigabit handoff from XO, ATT, Level 3 and Wiltel (I have a 100Mb didicated line). So I dont think its the Servers, its the network, Asterisk is working fine and all codecs look right...what could be the cause? **SNIP FROM SIP.CONF*** [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) ;bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records allow=ulaw ; Allow codecs in order of preference * ga0*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 64.201.99.2479092479878 2fd496bf330 00103/00105 ulaw ga0*CLI show version Asterisk 1.0.3 built by [EMAIL PROTECTED] on a x86_64 running Linux P.S. in my sip.conf file, it looks like I am only allowing the ulaw codec...could that cause a problem if I happen to need to call somebody that doesn't support ulaw? Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - sound still choppy
- Original Message - I see the sip user is an external ip. I would take a look at your QoS settings on your router. Make sure the voice traffic is getting the priority it deserves. Also, check for packet loss. I'd still be wondering if there's something else. I, too, experience choppy SIP connectivity from external IPs, but as I've mentioned in previous postings, I have a Vonage ATA that seems to have no problems keeping a crystal clear connection as it leaves my place and goes to Vonage's servers, so I think there must be more to it than QoS. I have to believe that there's some more jitter correction or other such buffering that could berhaps be played with, though I don't know what it would be *shrug*.? regards, Paul On Thu, 20 Jan 2005 11:26:02 -0800, Gabriel Afana [EMAIL PROTECTED] wrote: Hi, My SIP calls are sounding a little choppy. I've did my research but everything looks right on my end...what am I missing? Running RedHat ES 3.0 on dual AMD Opteron servers. My system is cololocated in downtown LA and is fed via a gigabit handoff from XO, ATT, Level 3 and Wiltel (I have a 100Mb didicated line). So I dont think its the Servers, its the network, Asterisk is working fine and all codecs look right...what could be the cause? **SNIP FROM SIP.CONF*** [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) ;bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records allow=ulaw ; Allow codecs in order of preference * ga0*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format 64.201.99.2479092479878 2fd496bf330 00103/00105 ulaw ga0*CLI show version Asterisk 1.0.3 built by [EMAIL PROTECTED] on a x86_64 running Linux P.S. in my sip.conf file, it looks like I am only allowing the ulaw codec...could that cause a problem if I happen to need to call somebody that doesn't support ulaw? Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
