Re: [Asterisk-Users] audio pause/delay problems
Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Jan Having investigated this a bit more, it turns out that using alaw Jan instead of gsm on the IAX2 link makes the problem go away. It Jan seems the jitter settings start working then. Jan Any hints? I'd prefer not to be stuck with 80kbps per call... Having investigated this further, it seems that connecting a zaptel device (WC100USB in my case) to the local * fixes the problem. --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users John ___ Asterisk-Users John mailing list [EMAIL PROTECTED] John http://lists.digium.com/mailman/listinfo/asterisk-users Jan ___ Asterisk-Users Jan mailing list [EMAIL PROTECTED] Jan http://lists.digium.com/mailman/listinfo/asterisk-users Jan ___ Asterisk-Users Jan mailing list [EMAIL PROTECTED] Jan http://lists.digium.com/mailman/listinfo/asterisk-users pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] audio pause/delay problems
I'm curious. Isn't anyone else noticing these problems? Or are people simply not using asterisk for VoIP connectivity over wide-area networks this way? Or does it go away with g729 or other proprietary codecs? --J. Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] audio pause/delay problems
I use IAX2 over a 2000mile loop from my home to the office using GSM and have no problems as long as the lag is low. Most of the time you can't tell the difference between VoIP and PSTN on the phones at home. On Mon, 2003-07-14 at 12:30, Jan Rychter wrote: I'm curious. Isn't anyone else noticing these problems? Or are people simply not using asterisk for VoIP connectivity over wide-area networks this way? Or does it go away with g729 or other proprietary codecs? --J. Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio pause/delay problems
This happens to me as I mention below, but only rarely. What is your CVS version? JT I'm curious. Isn't anyone else noticing these problems? Or are people simply not using asterisk for VoIP connectivity over wide-area networks this way? Or does it go away with g729 or other proprietary codecs? --J. Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. Content-Type: application/pgp-signature Attachment converted: PrivateSpace:Untitled 302 (/) (04203330) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio pause/delay problems
John == John Todd [EMAIL PROTECTED] writes: John This happens to me as I mention below, but only rarely. What is John your CVS version? The latest? E.g. I've tested 2 days ago. --J. I'm curious. Isn't anyone else noticing these problems? Or are people simply not using asterisk for VoIP connectivity over wide-area networks this way? Or does it go away with g729 or other proprietary codecs? --J. Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] audio pause/delay problems
Jan == Jan Rychter [EMAIL PROTECTED] writes: John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. Jan I have noticed these problems both in this kind of setup and in a Jan SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? Jan GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users John ___ Asterisk-Users John mailing list [EMAIL PROTECTED] John http://lists.digium.com/mailman/listinfo/asterisk-users Jan ___ Asterisk-Users Jan mailing list [EMAIL PROTECTED] Jan http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio pause/delay problems
John == John Todd [EMAIL PROTECTED] writes: John For what it's worth, I have noticed the same problem, but I think John the problem is in IAX2, since my long-haul portions of the John diagram were over IAX2, while my SIP clients are almost always John sitting on the same LAN as the Asterisk server. I have noticed these problems both in this kind of setup and in a SIP call to a remote Asterisk server. John What codec were you testing with over IAX2? GSM. --J. [I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP-PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on my side of the link. The setup then becomes: linphone - * - internet (IAX2) - * - PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users John ___ Asterisk-Users John mailing list [EMAIL PROTECTED] John http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
