Re: [Asterisk-Users] Sipura g729 call quality to PSTN
That does not sound right at all. The difference between the two Time= values should have been 10 (milliseconds). Did you reboot the Sipura after making the change? There are some values in the Sipura that don't take effect until after the next reboot; I don't have a clue whether this happens to be one of them. Yes - sipura was rebooted. Actually, the changes did seem to take affect even before the reboot (verified by call quality improvement and ethereal traces). So in your opinion, instead of 80, it should be a difference of 10? If so - then you are saying that the timestamp is in miliseconds? I am as puzzled as you - really does not seem logical, but call quality is finally decent and it does not seem to bother asterisk at all. Do you see any potential problems with this? I did a fair amount of experimenting this morning using a spa3000 with g711 and g729 codecs. I'm more confused now then ever. I also used ethereal to inspect timestamps, etc. spa3k(fxs) - asterisk - IAX(ITSP) - pstn net - analog phone The spa3k is running v2.0.13(GWg) firmware, and * is CVS-HEAD-02/13/05. The spa3k had a default RTP Packet Size of .030 (30 milliseconds) even though the User Manual indicated that 20 milliseconds is the default. Asterisk config is default at 20 milliseconds. I changed the spa3k rtp from .030 seconds, to .020 seconds for consistency. Audio quality seemed to be better when using g711. Regardless of whether I used g711u or g729, the rtp timestamps were always 160 difference between consequtive packets (as observed by ethereal). Changing the spa3k rtp to .010 seconds yielded timestamps that were always 80 difference between consequtive packets (same as you observed). However, * - spa3k continued to have 160 difference. Audio quality seemed to improve another step, and the occasional echo that we heard seemed to disappear. Pure guess is the smaller rtp size is impacting the jitter buffer and/or echo canceller in the spa3k. I'm going to run with these settings for a while to see what the longer term impact/stability might be. Rich ___ 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] Sipura g729 call quality to PSTN
Rich - thanks! Glad I am not the only one seeing this :) Would be very interested in your results. No problems that I see yet with these settings. On Fri, 18 Feb 2005 10:21:08 -0600, Rich Adamson [EMAIL PROTECTED] wrote: That does not sound right at all. The difference between the two Time= values should have been 10 (milliseconds). Did you reboot the Sipura after making the change? There are some values in the Sipura that don't take effect until after the next reboot; I don't have a clue whether this happens to be one of them. Yes - sipura was rebooted. Actually, the changes did seem to take affect even before the reboot (verified by call quality improvement and ethereal traces). So in your opinion, instead of 80, it should be a difference of 10? If so - then you are saying that the timestamp is in miliseconds? I am as puzzled as you - really does not seem logical, but call quality is finally decent and it does not seem to bother asterisk at all. Do you see any potential problems with this? I did a fair amount of experimenting this morning using a spa3000 with g711 and g729 codecs. I'm more confused now then ever. I also used ethereal to inspect timestamps, etc. spa3k(fxs) - asterisk - IAX(ITSP) - pstn net - analog phone The spa3k is running v2.0.13(GWg) firmware, and * is CVS-HEAD-02/13/05. The spa3k had a default RTP Packet Size of .030 (30 milliseconds) even though the User Manual indicated that 20 milliseconds is the default. Asterisk config is default at 20 milliseconds. I changed the spa3k rtp from .030 seconds, to .020 seconds for consistency. Audio quality seemed to be better when using g711. Regardless of whether I used g711u or g729, the rtp timestamps were always 160 difference between consequtive packets (as observed by ethereal). Changing the spa3k rtp to .010 seconds yielded timestamps that were always 80 difference between consequtive packets (same as you observed). However, * - spa3k continued to have 160 difference. Audio quality seemed to improve another step, and the occasional echo that we heard seemed to disappear. Pure guess is the smaller rtp size is impacting the jitter buffer and/or echo canceller in the spa3k. I'm going to run with these settings for a while to see what the longer term impact/stability might be. Rich ___ 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] Sipura g729 call quality to PSTN
Actually - jitter does not seem to be the issue (sound is not garbled and does not drop out, it was just very low and fuzzy/staticy when not set to 10 ms). It is weird that I have to drop to 10ms, but I have tested some more and the general consenses from the people I have called said it sounds fine now with 10ms setting. Thanks for your help though. Here is the result set from the ethereal trace using 10ms (RTP stream sent from Sipura to asterisk): RTP Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604121 RTP Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604201 As you can see there is now a difference of 80 between the Time stamps (now to sound dumb, but it would be 80 what?) On Wed, 16 Feb 2005 19:42:19 -0700, Keith Burns [EMAIL PROTECTED] wrote: Hmmm, that worked? Interesting that you can change the sample size to 10ms since the standard is 20ms that most people don't go below. I know you *can* do below 20 but if you are doubt the technical ability of the box it seems strange they are capable of that. This seems to smack of bad de-jitter buffers on the egress gateway... are you receiving 20ms sampled RTP ? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Wednesday, February 16, 2005 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN FYI - Seems the latest firmware in conjunction with changing the packet size to 10ms improved the call quality to usable. The Cisco 7960 is stell superior, but now at least the SPA-2100 is acceptable (and with 2 working g729 channels including 3-way calling). On Wed, 16 Feb 2005 15:44:58 -0500, Pedro [EMAIL PROTECTED] wrote: Forgot to mention that when I set the RTP Packet Size to 20ms that the difference was 160 (like the Cisco) but call quality was much worse. On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote: Thanks for the suggestion. Changing the RTP Packet Size in the Sipura to 40ms did improve the call quality slightly, but still well below par compared to the Cisco 7960. In my ethereal captures, I did notice something interesting. While the RTP stream from the Cisco to asterisk seemed to have a 160 diffference in timestamps, the Sipura showed a 320 difference: Cisco: RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 Sipura: RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns [EMAIL PROTECTED] wrote: What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
Actually - jitter does not seem to be the issue (sound is not garbled and does not drop out, it was just very low and fuzzy/staticy when not set to 10 ms). It is weird that I have to drop to 10ms, but I have tested some more and the general consenses from the people I have called said it sounds fine now with 10ms setting. Thanks for your help though. Here is the result set from the ethereal trace using 10ms (RTP stream sent from Sipura to asterisk): RTP Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604121 RTP Payload type=ITU-T G.729, SSRC=1783584165, Seq=7784, Time=156604201 As you can see there is now a difference of 80 between the Time stamps (now to sound dumb, but it would be 80 what?) That does not sound right at all. The difference between the two Time= values should have been 10 (milliseconds). Did you reboot the Sipura after making the change? There are some values in the Sipura that don't take effect until after the next reboot; I don't have a clue whether this happens to be one of them. ___ 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] Sipura g729 call quality to PSTN
That does not sound right at all. The difference between the two Time= values should have been 10 (milliseconds). Did you reboot the Sipura after making the change? There are some values in the Sipura that don't take effect until after the next reboot; I don't have a clue whether this happens to be one of them. Yes - sipura was rebooted. Actually, the changes did seem to take affect even before the reboot (verified by call quality improvement and ethereal traces). So in your opinion, instead of 80, it should be a difference of 10? If so - then you are saying that the timestamp is in miliseconds? I am as puzzled as you - really does not seem logical, but call quality is finally decent and it does not seem to bother asterisk at all. Do you see any potential problems with this? ___ 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] Sipura g729 call quality to PSTN
The only problem is that it is bandwidth inefficient and may cause a CPU hit on your IAD (since you have effectively doubled the pps for a call). The packets should be 10ms apart. Perhaps the timestamp is not in ms. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Thursday, February 17, 2005 5:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN That does not sound right at all. The difference between the two Time= values should have been 10 (milliseconds). Did you reboot the Sipura after making the change? There are some values in the Sipura that don't take effect until after the next reboot; I don't have a clue whether this happens to be one of them. Yes - sipura was rebooted. Actually, the changes did seem to take affect even before the reboot (verified by call quality improvement and ethereal traces). So in your opinion, instead of 80, it should be a difference of 10? If so - then you are saying that the timestamp is in miliseconds? I am as puzzled as you - really does not seem logical, but call quality is finally decent and it does not seem to bother asterisk at all. Do you see any potential problems with this? ___ 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] Sipura g729 call quality to PSTN
Thanks for the suggestion. Changing the RTP Packet Size in the Sipura to 40ms did improve the call quality slightly, but still well below par compared to the Cisco 7960. In my ethereal captures, I did notice something interesting. While the RTP stream from the Cisco to asterisk seemed to have a 160 diffference in timestamps, the Sipura showed a 320 difference: Cisco: RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 Sipura: RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns [EMAIL PROTECTED] wrote: What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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] Sipura g729 call quality to PSTN
Forgot to mention that when I set the RTP Packet Size to 20ms that the difference was 160 (like the Cisco) but call quality was much worse. On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote: Thanks for the suggestion. Changing the RTP Packet Size in the Sipura to 40ms did improve the call quality slightly, but still well below par compared to the Cisco 7960. In my ethereal captures, I did notice something interesting. While the RTP stream from the Cisco to asterisk seemed to have a 160 diffference in timestamps, the Sipura showed a 320 difference: Cisco: RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 Sipura: RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns [EMAIL PROTECTED] wrote: What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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] Sipura g729 call quality to PSTN
FYI - Seems the latest firmware in conjunction with changing the packet size to 10ms improved the call quality to usable. The Cisco 7960 is stell superior, but now at least the SPA-2100 is acceptable (and with 2 working g729 channels including 3-way calling). On Wed, 16 Feb 2005 15:44:58 -0500, Pedro [EMAIL PROTECTED] wrote: Forgot to mention that when I set the RTP Packet Size to 20ms that the difference was 160 (like the Cisco) but call quality was much worse. On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote: Thanks for the suggestion. Changing the RTP Packet Size in the Sipura to 40ms did improve the call quality slightly, but still well below par compared to the Cisco 7960. In my ethereal captures, I did notice something interesting. While the RTP stream from the Cisco to asterisk seemed to have a 160 diffference in timestamps, the Sipura showed a 320 difference: Cisco: RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 Sipura: RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns [EMAIL PROTECTED] wrote: What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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] Sipura g729 call quality to PSTN
Title: RE: [Asterisk-Users] Sipura g729 call quality to PSTN Next thing I would check are the de-jitter buffers if possible on the Sipura, or jitter in general. Do you have control of the PSTN gateway ? Measure the jitter on ingress to the gateway. You can do this crudely by using Ethereal and looking at the delta between timestamps on RTP packets from Sipura to PSTN. -Original Message- From: Pedro [mailto:[EMAIL PROTECTED]] Sent: Wednesday, February 16, 2005 1:37 PM To: Keith Burns Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Jeffrey Chan Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Thanks for the suggestion. Changing the RTP Packet Size in the Sipura to 40ms did improve the call quality slightly, but still well below par compared to the Cisco 7960. In my ethereal captures, I did notice something interesting. While the RTP stream from the Cisco to asterisk seemed to have a 160 diffference in timestamps, the Sipura showed a 320 difference: Cisco: RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 Sipura: RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns [EMAIL PROTECTED] wrote: What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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] Sipura g729 call quality to PSTN
Hmmm, that worked? Interesting that you can change the sample size to 10ms since the standard is 20ms that most people don't go below. I know you *can* do below 20 but if you are doubt the technical ability of the box it seems strange they are capable of that. This seems to smack of bad de-jitter buffers on the egress gateway... are you receiving 20ms sampled RTP ? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Wednesday, February 16, 2005 3:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN FYI - Seems the latest firmware in conjunction with changing the packet size to 10ms improved the call quality to usable. The Cisco 7960 is stell superior, but now at least the SPA-2100 is acceptable (and with 2 working g729 channels including 3-way calling). On Wed, 16 Feb 2005 15:44:58 -0500, Pedro [EMAIL PROTECTED] wrote: Forgot to mention that when I set the RTP Packet Size to 20ms that the difference was 160 (like the Cisco) but call quality was much worse. On Wed, 16 Feb 2005 15:36:44 -0500, Pedro [EMAIL PROTECTED] wrote: Thanks for the suggestion. Changing the RTP Packet Size in the Sipura to 40ms did improve the call quality slightly, but still well below par compared to the Cisco 7960. In my ethereal captures, I did notice something interesting. While the RTP stream from the Cisco to asterisk seemed to have a 160 diffference in timestamps, the Sipura showed a 320 difference: Cisco: RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57094, Time=40666896 RTP Payload type=ITU-T G.729, SSRC=1794532067, Seq=57095, Time=40667056 Sipura: RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434932771 RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784, Time=434933091 On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns [EMAIL PROTECTED] wrote: What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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
Re: [Asterisk-Users] Sipura g729 call quality to PSTN
On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Sipura g729 call quality to PSTN
uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Sipura g729 call quality to PSTN
Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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] Sipura g729 call quality to PSTN
What is your sample size? I believe the 7960 supports 40ms (2 samples) per packet by default. Do you have an ethereal trace? Look at the timestamps between RTP packets if you can't see/modify this setting. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Pedro Sent: Tuesday, February 15, 2005 6:30 PM To: Jeffrey Chan Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura g729 call quality to PSTN Actually the SPA-2100 supports 2 g729 channels which is why I bought it. Unfortunately, the call quality is just as poor on the 2100 as it is on the 2000. - Pedro On Tue, 15 Feb 2005 23:12:51 +, Jeffrey Chan [EMAIL PROTECTED] wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess too . why SPA-2000 supports G729 for one channel only? no enough CPU power to code/decode G.729 for two channels? Jeffey www.mutualphone.com On Tue, 15 Feb 2005 16:31:59 -0500, Pedro [EMAIL PROTECTED] wrote: uggg. Is anyone out there having any luck with the SPA-2000 or SPA-2100 using the g729 codec with decent call quality? On Tue, 15 Feb 2005 10:19:05 -0500, Mark Eissler [EMAIL PROTECTED] wrote: On Feb 14, 2005, at 1:25 PM, Pedro wrote: Is it just a bad implementation of g729 compression with the Sipura product line? That would be my guess. -mark -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ 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 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] Sipura g729 call quality to PSTN
If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side (the audio on the phone connected to the Sipura sounds fine). My guess is that the Sipura does not compress the outbound audio very effectively and since the incoming audio from the PSTN is already compressed by the VoIP provider, it is just delivering the good-sounding g729 stream. It is worth noting that call quality on both the IP and PSTN side is great when using the Cisco 7960 with g729. It is just with the Sipura that the sound quality on the PSTN-side sounds like a bad quality cell phone call. I even got an SPA-2100 in hopes that the g729 would sound better on that unit, but the same issue is present there as well. Is it just a bad implementation of g729 compression with the Sipura product line? Any thoughts or recommendations are appreciated :) Thanks! - Pedro ___ 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
