RE: [Asterisk-Users] Re: Polycom sound quality problems
There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw on SIP to the phone. I considered that as a possibility originally, and even tried using GSM with Sixtel to force it to do transcoding, but had the exact same problem. The Asterisk box is a 2.4ghz P4 with 512MB RAM, doing nothing but Asterisk. I have only 9 extensions. That should not be a problem with those specs. I would think there's a possibility of packet loss on the IAX channel, except the other SIP phones (SJPhone softphone) work flawlessly. Also, OUTBOUND calls are just fine on the Polycoms. Only incoming calls are messed up. There is one other possibility. Are you using Switches or hubs? if you are using hubs you could have collisions that cause data loss. Switches with store and forward are the best especially if they have QoS, ToS and CoS management features in the switch. Switches typically do not loose packets but they do expire on rare occasions due to high traffic volume. Max ___ 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] Re: Polycom sound quality problems
Interesting news... I just got a call from one of the SIP phones outside our LAN, over a VPN, with reinvite disabled, and it sounded like a robot. Calls from SIP phones on the VPN sound fine when reinvite is enabled. So it seems ANY call Asterisk bridges to the Polycom sounds crappy. Maybe this will shed some light on the issue. Eric Noah Miller wrote: There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw on SIP to the phone. I considered that as a possibility originally, and even tried using GSM with Sixtel to force it to do transcoding, but had the exact same problem. The Asterisk box is a 2.4ghz P4 with 512MB RAM, doing nothing but Asterisk. I have only 9 extensions. I would think there's a possibility of packet loss on the IAX channel, except the other SIP phones (SJPhone softphone) work flawlessly. Also, OUTBOUND calls are just fine on the Polycoms. Only incoming calls are messed up. Just to cover all the bases, have you tried any other IAX providers or connections? ___ 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] Re: Polycom sound quality problems
There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw on SIP to the phone. I considered that as a possibility originally, and even tried using GSM with Sixtel to force it to do transcoding, but had the exact same problem. The Asterisk box is a 2.4ghz P4 with 512MB RAM, doing nothing but Asterisk. I have only 9 extensions. I would think there's a possibility of packet loss on the IAX channel, except the other SIP phones (SJPhone softphone) work flawlessly. Also, OUTBOUND calls are just fine on the Polycoms. Only incoming calls are messed up. Max W Blackmer Jr wrote: I don't see any way to tell the Polycom to ignore QoS. It's mainly routers and switches that pay attention to QoS, the phone would just set QoS on its outgoing packets. Anyway, here's what's in the QoS section- it all seems to be related to sending packets: It is not in the transport if it is sounding bad look and see if there is any transcoding occuring from the IAX to the SIP. What codecs are accepted on the AIX should be the Same codecs accepted on the SIP channel ... and what codects are being used on each phone. This sounds like a transcoding issue. ___ 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
[Asterisk-Users] Re: Polycom sound quality problems
There's no transcoding going on. It's ulaw on IAX with Sixtel and ulaw on SIP to the phone. I considered that as a possibility originally, and even tried using GSM with Sixtel to force it to do transcoding, but had the exact same problem. The Asterisk box is a 2.4ghz P4 with 512MB RAM, doing nothing but Asterisk. I have only 9 extensions. I would think there's a possibility of packet loss on the IAX channel, except the other SIP phones (SJPhone softphone) work flawlessly. Also, OUTBOUND calls are just fine on the Polycoms. Only incoming calls are messed up. Just to cover all the bases, have you tried any other IAX providers or connections? ___ 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] Re: Polycom sound quality problems
Hi Eric - I'm having a problem with my Polycom phones and hoping someone else has experienced the same thing: Outbound calls are fine, and inbound calls originating from another SIP phone are fine, but inbound calls to the Polycom phone from an IAX channel sound like you're talking to a robot. The person on the Polycom sounds fine to the person on the IAX channel, however. Inbound calls to our soft phones sound just fine. Asterisk 1.0.5 on Debian (also had the problem with 1.0 on Fedora) Polycom SoundPoint IP500 SIP Sixtel is the IAX provider. Check to see what codec is being used for the call. Sean Default is U-law, but I also switched it to A-law with the exact same results. I might check out QoS. You can specify TOS tagging on your IAX channels in iax.conf, and the Polycom phones are able to respond to TOS tagging (in ipmid.cfg - or in the web interface under Core Conf). Maybe they are are trying to do two mutually exclusive kinds of TOS tagging? You can tell the Polycom phone to just not respond to TOS. - Noah ___ 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] Re: Polycom sound quality problems
How can TOS tagging on the IAX channel affect a phone that is completely SIP? In my experience, the robot voice issue usually arises when bandwidth restriction or latency occur on the data line providing the IAX call. Assuming your connection is like this Sixtel IAX --- Asterisk Box --- Polycom IP500 Then you can see that the latency problem is on the IAX side. What is your setup? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Friday, April 01, 2005 6:44 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Polycom sound quality problems Hi Eric - I'm having a problem with my Polycom phones and hoping someone else has experienced the same thing: Outbound calls are fine, and inbound calls originating from another SIP phone are fine, but inbound calls to the Polycom phone from an IAX channel sound like you're talking to a robot. The person on the Polycom sounds fine to the person on the IAX channel, however. Inbound calls to our soft phones sound just fine. Asterisk 1.0.5 on Debian (also had the problem with 1.0 on Fedora) Polycom SoundPoint IP500 SIP Sixtel is the IAX provider. Check to see what codec is being used for the call. Sean Default is U-law, but I also switched it to A-law with the exact same results. I might check out QoS. You can specify TOS tagging on your IAX channels in iax.conf, and the Polycom phones are able to respond to TOS tagging (in ipmid.cfg - or in the web interface under Core Conf). Maybe they are are trying to do two mutually exclusive kinds of TOS tagging? You can tell the Polycom phone to just not respond to TOS. - Noah ___ 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
[Asterisk-Users] Re: Polycom sound quality problems
I don't see any way to tell the Polycom to ignore QoS. It's mainly routers and switches that pay attention to QoS, the phone would just set QoS on its outgoing packets. Anyway, here's what's in the QoS section- it all seems to be related to sending packets: QoS RTP 802.1Q User Priority IP ToS Minimize Delay Enabled Disabled IP ToS Maximize Throughput Enabled Disabled IP ToS Maximize Reliability Enabled Disabled IP ToS Minimize Cost Enabled Disabled IP ToS Precedence Call Control 802.1Q User Priority IP ToS Minimize Delay Enabled Disabled IP ToS Maximize Throughput Enabled Disabled IP ToS Maximize Reliability Enabled Disabled IP ToS Minimize Cost Enabled Disabled IP ToS Precedence Other Protocols 802.1Q User Priority The problem is not that it's choppy or breaks up. Asterisk is connected to the phone through two 100mbit switches, so throughput isn't a problem. It just sounds very distorted, like a cross between a robot and Donald Duck. It really seems to be a problem with the way Asterisk is bridging the call from IAX to the phone. It does SIP - SIP bridges (not reinviting) just fine. Noah Miller wrote: Hi Eric - I'm having a problem with my Polycom phones and hoping someone else has experienced the same thing: Outbound calls are fine, and inbound calls originating from another SIP phone are fine, but inbound calls to the Polycom phone from an IAX channel sound like you're talking to a robot. The person on the Polycom sounds fine to the person on the IAX channel, however. Inbound calls to our soft phones sound just fine. Asterisk 1.0.5 on Debian (also had the problem with 1.0 on Fedora) Polycom SoundPoint IP500 SIP Sixtel is the IAX provider. Check to see what codec is being used for the call. Sean Default is U-law, but I also switched it to A-law with the exact same results. I might check out QoS. You can specify TOS tagging on your IAX channels in iax.conf, and the Polycom phones are able to respond to TOS tagging (in ipmid.cfg - or in the web interface under Core Conf). Maybe they are are trying to do two mutually exclusive kinds of TOS tagging? You can tell the Polycom phone to just not respond to TOS. - Noah ___ 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] Re: Polycom sound quality problems
See my other email My setup VoIP Provider --- My T1 --- Asterisk --- Sip Clients The only time I get robo-voice is when the latency to the VoIP provider is high. Translating from IAX to SIP should not be a problem but maybe it is in the build you have? I run COS on my Polycom segment but with 100 meg switches (9GB Backplane) and 100 MB network, there is little internal latency. I would look to the external IAX segment. W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Mason Sent: Friday, April 01, 2005 9:22 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Polycom sound quality problems I don't see any way to tell the Polycom to ignore QoS. It's mainly routers and switches that pay attention to QoS, the phone would just set QoS on its outgoing packets. Anyway, here's what's in the QoS section- it all seems to be related to sending packets: QoS RTP 802.1Q User Priority IP ToS Minimize Delay Enabled Disabled IP ToS Maximize Throughput Enabled Disabled IP ToS Maximize Reliability Enabled Disabled IP ToS Minimize Cost Enabled Disabled IP ToS Precedence Call Control 802.1Q User Priority IP ToS Minimize Delay Enabled Disabled IP ToS Maximize Throughput Enabled Disabled IP ToS Maximize Reliability Enabled Disabled IP ToS Minimize Cost Enabled Disabled IP ToS Precedence Other Protocols 802.1Q User Priority The problem is not that it's choppy or breaks up. Asterisk is connected to the phone through two 100mbit switches, so throughput isn't a problem. It just sounds very distorted, like a cross between a robot and Donald Duck. It really seems to be a problem with the way Asterisk is bridging the call from IAX to the phone. It does SIP - SIP bridges (not reinviting) just fine. Noah Miller wrote: Hi Eric - I'm having a problem with my Polycom phones and hoping someone else has experienced the same thing: Outbound calls are fine, and inbound calls originating from another SIP phone are fine, but inbound calls to the Polycom phone from an IAX channel sound like you're talking to a robot. The person on the Polycom sounds fine to the person on the IAX channel, however. Inbound calls to our soft phones sound just fine. Asterisk 1.0.5 on Debian (also had the problem with 1.0 on Fedora) Polycom SoundPoint IP500 SIP Sixtel is the IAX provider. Check to see what codec is being used for the call. Sean Default is U-law, but I also switched it to A-law with the exact same results. I might check out QoS. You can specify TOS tagging on your IAX channels in iax.conf, and the Polycom phones are able to respond to TOS tagging (in ipmid.cfg - or in the web interface under Core Conf). Maybe they are are trying to do two mutually exclusive kinds of TOS tagging? You can tell the Polycom phone to just not respond to TOS. - Noah ___ 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
[Asterisk-Users] Re: Polycom sound quality problems
How can TOS tagging on the IAX channel affect a phone that is completely SIP? Quite right, Wiley. I think I need my head checked. ___ 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] Re: Polycom sound quality problems
I don't see any way to tell the Polycom to ignore QoS. It's mainly routers and switches that pay attention to QoS, the phone would just set QoS on its outgoing packets. Anyway, here's what's in the QoS section- it all seems to be related to sending packets: It is not in the transport if it is sounding bad look and see if there is any transcoding occuring from the IAX to the SIP. What codecs are accepted on the AIX should be the Same codecs accepted on the SIP channel ... and what codects are being used on each phone. This sounds like a transcoding issue. ___ 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] Re: Polycom sound quality problems
Has the users hardware been assessed yet? I cannot remember seing anything regarding the hardware for this issue. I am sure memory and processor speed will play a part if lots of calls are active during the transcode... W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max W Blackmer Jr Sent: Friday, April 01, 2005 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Re: Polycom sound quality problems I don't see any way to tell the Polycom to ignore QoS. It's mainly routers and switches that pay attention to QoS, the phone would just set QoS on its outgoing packets. Anyway, here's what's in the QoS section- it all seems to be related to sending packets: It is not in the transport if it is sounding bad look and see if there is any transcoding occuring from the IAX to the SIP. What codecs are accepted on the AIX should be the Same codecs accepted on the SIP channel ... and what codects are being used on each phone. This sounds like a transcoding issue. ___ 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
