Re: [asterisk-users] SIP Delay with remote stations?
William Stillwell (Lists) writes: I have several remote phones that experience a slight call delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? This is actually a somewhat common problem in SIP. One end sends media before the other end is ready to receive it, or a gateway receives media on one leg of the call but media isn't yet ready on the other leg... In your case I would guess that it is caused by firewalls/NAT reacting only to RTP traffic in one direction, thereby blocking traffic in the other until the first packet. Luckily it's IP, so you can use tcpdump or wireshark or phone-specific dump tools to capture the traffic and see where the problem hides. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Delay with remote stations?
this can be cause if you are using an ADSL link with your remote phones .. or maybe some 3G networks can cause that delay in the first response as the ACK message will be late to arrive and if the delay was too high .. the call will drop.one more thing if your remote phones are (Queue Members) this can be caused by a configuration of the queue itself something related to memberdelay directive. try setting it to 0 or something similar.Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 386 492 9993 From: william.stillwell-li...@ablebody.net To: asterisk-users@lists.digium.com Date: Tue, 29 Jun 2010 10:06:55 -0400 Subject: [asterisk-users] SIP Delay with remote stations? I have several remote phones that experience a slight “call” delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. _ The New Busy think 9 to 5 is a cute idea. Combine multiple calendars with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multicalendarocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_5-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Delay with remote stations?
I have several remote phones that experience a slight call delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Delay with remote stations?
On Tue, Jun 29, 2010 at 10:06 AM, William Stillwell (Lists) william.stillwell-li...@ablebody.net wrote: Any idea what could be causing this? Yes, network delay, packet loss, the Internet. Implement QoS and bandwidth monitoring. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
