Hi guys, I have the following configuration: 10 SIP softphones <--> Asterisk <--> PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex
If the PSTN side is only complaining about conversations with a single phone on the SIP side, look at the SIP phone. Check the settings for that SIP phone/PC (VAD disabled, NIC settings, runaway processes). Do PSTN callers here choppiness from the SIP phone caller? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Asterisk Sent: Monday, February 12, 2007 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bad audio quality on SIP Hi guys, I have the following configuration: 10 SIP softphones <--> Asterisk <--> PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex _______________________________________________ --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
Well, the PSTN side is complaining about a random phone on the SIP side. Yes, they do hear choppiness. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, February 12, 2007 2:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bad audio quality on SIP If the PSTN side is only complaining about conversations with a single phone on the SIP side, look at the SIP phone. Check the settings for that SIP phone/PC (VAD disabled, NIC settings, runaway processes). Do PSTN callers here choppiness from the SIP phone caller? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Asterisk Sent: Monday, February 12, 2007 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bad audio quality on SIP Hi guys, I have the following configuration: 10 SIP softphones <--> Asterisk <--> PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex _______________________________________________ --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
It's all in the local LAN network - client computers (with SIP softphones) are connected and registered at Asterisk SIP proxy via 100 MB connection each. The QoS is enabled under TCP/IP protocol in LAN connection in Windows (cause SIP softphones are running in Windows environment), and tos in sip.conf is set to 0x18. Unfortunately I don't have access to switch to tell you how it's set up there, but the network technicians said it is enabled. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, February 12, 2007 3:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bad audio quality on SIP If it's a random phone on the SIP side, we have to look further upstream. While jitterbuffers may help, in my opinion they mask a problem. What type of connection do you have to the internet? Have you done tracert's to your voip provider? What do they look like? When you say that you do QoS - how? What device and settings/app helper? MD -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Asterisk Sent: Monday, February 12, 2007 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bad audio quality on SIP Well, the PSTN side is complaining about a random phone on the SIP side. Yes, they do hear choppiness. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Monday, February 12, 2007 2:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bad audio quality on SIP If the PSTN side is only complaining about conversations with a single phone on the SIP side, look at the SIP phone. Check the settings for that SIP phone/PC (VAD disabled, NIC settings, runaway processes). Do PSTN callers here choppiness from the SIP phone caller? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Asterisk Sent: Monday, February 12, 2007 8:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bad audio quality on SIP Hi guys, I have the following configuration: 10 SIP softphones <--> Asterisk <--> PSTN Audio is always good on SIP softphone side, but callers from PSTN side *sometimes* complain that the audio quality is bad (and volume low). The QoS is turned on on the computers where SIP softphone is installed, and the tos setting in sip.conf is set to 0x18. The interesting thing is that usually only one SIP softphone user (but each time someone else) would get complaints like that ... others seem to work okay. What could be wrong? Thanx, Alex _______________________________________________ --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 _______________________________________________ --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