Re: [Asterisk-Users] How to increase the performance?
In the past I had problems with the audio over sip. Then I tried the -p Option and increased the memory. Now it is better but not perfect. Are there any more possibilities to increase it more? By now I'm using a P-II/333. Could a completely hand optimized kernel (I use 2.6.) help a bit? There's no way to answer your question with any degree of reasonable truth as you haven't mentioned they type of phones, type of pstn interface, which codecs, etc, etc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to increase the performance?
Rich Adamson schrieb: In the past I had problems with the audio over sip. Then I tried the -p Option and increased the memory. Now it is better but not perfect. Are there any more possibilities to increase it more? By now I'm using a P-II/333. Could a completely hand optimized kernel (I use 2.6.) help a bit? There's no way to answer your question with any degree of reasonable truth as you haven't mentioned they type of phones, type of pstn interface, which codecs, etc, etc. Okay. My server has got: - One Phonejack Lite - One X100P Clone - 256mb Memory - P II/333 - Linux 2.6.5 - Debian Woody - Asterisk 1.0.1 - Codecs: GSM, ulaw, alaw - ADSL 1000kBit/s Downstream, 128kBit/s Upstream Calls from or to the pstn are completely okay. Calls over SIP aren't. Calls over IAX couldn't be tested at the moment. Do you need any more facts? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to increase the performance?
In the past I had problems with the audio over sip. Then I tried the -p Option and increased the memory. Now it is better but not perfect. Are there any more possibilities to increase it more? By now I'm using a P-II/333. Could a completely hand optimized kernel (I use 2.6.) help a bit? There's no way to answer your question with any degree of reasonable truth as you haven't mentioned they type of phones, type of pstn interface, which codecs, etc, etc. Okay. My server has got: - One Phonejack Lite - One X100P Clone - 256mb Memory - P II/333 - Linux 2.6.5 - Debian Woody - Asterisk 1.0.1 - Codecs: GSM, ulaw, alaw - ADSL 1000kBit/s Downstream, 128kBit/s Upstream Calls from or to the pstn are completely okay. Calls over SIP aren't. Calls over IAX couldn't be tested at the moment. Do you need any more facts? Sure, getting closer... Help us understand what calls over sip means. From what device to what device when the call is bad (need to understand the path that you're talking about including any transcoding going on (if any), what type of sip phone, is the sip connection local or through the dsl, and the other end of this 'bad call' where is it? Current version of * or what? Where ever this sip connection goes that is you're referring to, are there any CLI errors or have you tried to use a packet sniffer to see what's going on? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to increase the performance?
[EMAIL PROTECTED] wrote: Rich Adamson schrieb: In the past I had problems with the audio over sip. Then I tried the -p Option and increased the memory. Now it is better but not perfect. Are there any more possibilities to increase it more? By now I'm using a P-II/333. Could a completely hand optimized kernel (I use 2.6.) help a bit? There's no way to answer your question with any degree of reasonable truth as you haven't mentioned they type of phones, type of pstn interface, which codecs, etc, etc. Okay. My server has got: - One Phonejack Lite - One X100P Clone - 256mb Memory - P II/333 - Linux 2.6.5 - Debian Woody - Asterisk 1.0.1 - Codecs: GSM, ulaw, alaw - ADSL 1000kBit/s Downstream, 128kBit/s Upstream That upstream bandwitch will need to be managed carefully. If you're using G.711, one channel would be using roughly 80kbit of your upstream. Who has the most quality complaints: you, or the people you are talking to? Calls from or to the pstn are completely okay. Calls over SIP aren't. Calls over IAX couldn't be tested at the moment. Can you make a SIP connection directly to the box? No LAN, no WAN, just a crossover cable between your SIP phone (soft or hard) and your Asterisk system? That'll give us some idea of whether the problem is network or server-based. Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to increase the performance?
Rich Adamson schrieb: Do you need any more facts? Sure, getting closer... Help us understand what calls over sip means. From what device to what device when the call is bad (need to understand the path that you're talking about including any transcoding going on (if any), what type of sip phone, is the sip connection local or through the dsl, and the other end of this 'bad call' where is it? Okay. The call goes to or from a phone connected to the Phonejack Lite. The Asterisk Server (Version 1.0.1) converts from signed linear (Phonejack) to ULAW/ALAW/GSM. Then it is transferred over ADSL to sipgate.de (ping time 100ms) (most sip-calls are incoming calls from the pstn to the sipgate.de-gateway) Current version of * or what? No. It's the 1.0.1 since there are no newer versions available for debian woody. Where ever this sip connection goes that is you're referring to, are there any CLI errors or have you tried to use a packet sniffer to see what's going on? I haven't used any packet sniffer by now. The sound problems are occuring in 10-30% of the time. So I guess this can only be a performance or traffic problem. But since the sound problems are occuring in both directions I guess its a performance problem. Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to increase the performance?
Jim Van Meggelen schrieb: [EMAIL PROTECTED] wrote: - ADSL 1000kBit/s Downstream, 128kBit/s Upstream That upstream bandwitch will need to be managed carefully. I know. If you're using G.711, one channel would be using roughly 80kbit of your upstream. Who has the most quality complaints: you, or the people you are talking to? I don't know. I guess its equal. Since the problems occur even when I'm using GSM the bandwith shouldn't be the problem. Calls from or to the pstn are completely okay. Calls over SIP aren't. Calls over IAX couldn't be tested at the moment. Can you make a SIP connection directly to the box? No LAN, no WAN, just a crossover cable between your SIP phone (soft or hard) and your Asterisk system? No. My phone is connected via the Phonejack in the server that runs asterisk. But ... I could try to setup a software client that I connect to the server. Bye! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to increase the performance?
[EMAIL PROTECTED] wrote: Jim Van Meggelen schrieb: [EMAIL PROTECTED] wrote: - ADSL 1000kBit/s Downstream, 128kBit/s Upstream That upstream bandwitch will need to be managed carefully. I know. If you're using G.711, one channel would be using roughly 80kbit of your upstream. Who has the most quality complaints: you, or the people you are talking to? I don't know. I guess its equal. Since the problems occur even when I'm using GSM the bandwith shouldn't be the problem. Not as much, but if you have any other outbound traffic over that connection, it WILL still be a factor. Have you got a firewall/router that provides any QoS features? Calls from or to the pstn are completely okay. Calls over SIP aren't. Calls over IAX couldn't be tested at the moment. Can you make a SIP connection directly to the box? No LAN, no WAN, just a crossover cable between your SIP phone (soft or hard) and your Asterisk system? No. My phone is connected via the Phonejack in the server that runs asterisk. But ... I could try to setup a software client that I connect to the server. That might help to determine if the problem is in the Asterisk box or the network. If you can make a local SIP client talk to your Phonejack cleanly, and yet have problems with network connection, that could be a factor. Also, I recommend getting an account on IAXtel or FWD, and connecting a softphone directly to those networks, and then into your system. That'll stress your outgoing bandwidth, and potentially provide all kinds of interesting information about what works and doesn't. Cheers, Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] How to increase the performance?
Do you need any more facts? Sure, getting closer... Help us understand what calls over sip means. From what device to what device when the call is bad (need to understand the path that you're talking about including any transcoding going on (if any), what type of sip phone, is the sip connection local or through the dsl, and the other end of this 'bad call' where is it? Okay. The call goes to or from a phone connected to the Phonejack Lite. The Asterisk Server (Version 1.0.1) converts from signed linear (Phonejack) to ULAW/ALAW/GSM. Then it is transferred over ADSL to sipgate.de (ping time 100ms) (most sip-calls are incoming calls from the pstn to the sipgate.de-gateway) Current version of * or what? No. It's the 1.0.1 since there are no newer versions available for debian woody. Where ever this sip connection goes that is you're referring to, are there any CLI errors or have you tried to use a packet sniffer to see what's going on? I haven't used any packet sniffer by now. The sound problems are occuring in 10-30% of the time. So I guess this can only be a performance or traffic problem. But since the sound problems are occuring in both directions I guess its a performance problem. For sip calls, asterisk essentially obtains its timing from the remote sip device. If packets are dropped or missing for whatever reason between the sip device and asterisk, audio will be impacted in both directions (usually). Given the small bandwidth noted for your dsl connection, it is entirely possible that is the problem. But, you can verify that fairly easily by using a packet sniffer (eg, Ethereal) and looking at a few timestamps within the rtp packets. The timestamps should be consistently increasing by exactly the same amount (for both directions). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
