search for: rtt

...t smb service response times over OpenVPN using Wireshark/TShark. Here are 2 samples. In both samples a Win XP Pro machine is connecting to a server running samba-3.3.7 and openvpn-2.0.6_9 on FreeBSD 7.2-STABLE. Location #1 =================================================================== SMB RTT Statistics: Filter: Commands Calls Min RTT Max RTT Avg RTT Close 671 0.12442 0.90853 0.14041 Write 1 0.12672 0.12672 0.12672 Trans 11 0.12736 0.15758 0.13410 Echo...

On 15.05.20 at 14:31 Doug Lytle wrote: > Google says Round Trip Time > > https://www.voip-info.org/asterisk-rtcp/ That doesn't answer my question (I know the abbreviation RTT). Therefore I'm trying again: I'm just wondering what the RTT *exactly* means. Where are the exact measuring points located? => How are the RTT values exactly calculated? Which values are actually used for? => What about the processing time between the inbound leg and the outbound...

...o 0.15) on the FreeBSD machines. Here is the output from hping2 (excuse me the line wrap), root:~# hping -S -p 80 -c 6 www.freebsd.org HPING www.freebsd.org (fxp0 216.136.204.117): S set, 40 headers + 0 data bytes len=46 ip=216.136.204.117 ttl=50 DF id=20997 sport=80 flags=SA seq=0 win=57344 rtt=212.6 ms len=46 ip=216.136.204.117 ttl=50 DF id=21140 sport=80 flags=SA seq=1 win=57344 rtt=212.3 ms len=46 ip=216.136.204.117 ttl=50 DF id=21301 sport=80 flags=SA seq=2 win=57344 rtt=212.3 ms DUP! len=46 ip=216.136.204.117 ttl=50 DF id=21494 sport=80 flags=SA seq=0 win=57344 rtt=3207.3 ms len=...

Could you offer some suggestion how to fetch mailbox content over high RTT link (with negligible packet loss)? Currently I use IMAP+IDLE *but* it fails to use full available bandwidth due to high RTT and "send command wait for response" nature of POP3 and IMAP4 protocols. -- [pl>en: Andrew] Andrzej Adam Filip : anfi at onet.eu Learning without thought is l...

Hello! I'm just wondering what the RTT exactly means. Where are the exact measuring points located? > pjsip show channelstats ...........Receive......... .........Transmit.......... BridgeId ChannelId ........ UpTime.. Codec. Count Lost Pct Jitter Count Lost Pct Jitter RTT...

Hello all, We would like to pull the RTT of registered endpoints from MySQL for use in a webportal. However it doesn't appear asterisk tables this by default like chan_sip did. I've found some information [1] that a modification of sorcery.conf can get it writing to a table. But I'm struggling to figure out what that configur...

...n Sat, May 16, 2020 at 10:58 AM Michael Maier <m1278468 at mailbox.org> wrote: > On 15.05.20 at 14:31 Doug Lytle wrote: > > Google says Round Trip Time > > > > https://www.voip-info.org/asterisk-rtcp/ > > That doesn't answer my question (I know the abbreviation RTT). Therefore > I'm trying again: > > I'm just wondering what the RTT *exactly* means. Where are the exact > measuring points located? > > => How are the RTT values exactly calculated? Which values are actually > used for? > The value is calculated according to the...

On 17.05.20 at 01:28 Joshua C. Colp wrote: > On Sat, May 16, 2020 at 10:58 AM Michael Maier <m1278468 at mailbox.org> wrote: > >> => How are the RTT values exactly calculated? Which values are actually >> used for? >> > > The value is calculated according to the logic in the RFC[1]. Specifically > using embedded timestamps in the RTCP packets and calculated delays. The > value is presented in seconds I believe in the ou...

...conf ----------- [general] ; autofill=yes keepstats=yes setinterfacevar=yes ; [Kinna] retry=5 ringinuse=no strategy=rrmemory timeout=20 wrapuptime=120 extensions.conf --------------- exten => 0320209030,1,Answer() exten => 0320209030,n,ExecIf($[${QUEUE_MEMBER(Kinna,logged)}=0]?Queue(Goteborg,rtT)) exten => 0320209030,n,Queue(Kinna,nrtT) exten => 0320209030,n,NoOp(${MEMBERINTERFACE}) exten => 0320209030,n,NoOp(${MEMBERNAME}) exten => 0320209030,n,Queue(Goteborg,rtT) exten => 0320209030,n,Hangup() CLI> ---- == Using SIP RTP CoS mark 5 -- Executing [0320209030 at inpu...

...21-408-5000 x103 Mobile: 321-794-0763 On Tue, May 26, 2020 at 2:57 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, May 26, 2020 at 10:48 AM Nick Olsen < > nick at floridavirtualsolutions.com> wrote: > >> Hello all, >> >> We would like to pull the RTT of registered endpoints from MySQL for use >> in a webportal. However it doesn't appear asterisk tables this by default >> like chan_sip did. >> >> I've found some information [1] that a modification of sorcery.conf can >> get it writing to a table. But I'm...

On Tue, May 26, 2020 at 10:48 AM Nick Olsen < nick at floridavirtualsolutions.com> wrote: > Hello all, > > We would like to pull the RTT of registered endpoints from MySQL for use > in a webportal. However it doesn't appear asterisk tables this by default > like chan_sip did. > > I've found some information [1] that a modification of sorcery.conf can > get it writing to a table. But I'm struggling to figur...

Hi, I need a feature from asterisk with dahdi channels, if there is an incoming call, it should ring on several dahdi channels. My channels look like: OFFICE1=DAHDI/13,,rtT OFFICE2=DAHDI/14,,rtT If I add this line: exten => 12345678,1,Dial(${OFFICE1}&{OFFICE2}) only OFFICE1 rings. If I change it to exten => 12345678,1,Dial(DAHDI/13&DAHDI/14) DAHDI/13 and 14 rings together, but I can't add the ,,rtT features (the biggest problem is the person w...

I am using CHANNEL function to store the rtt as variable in cdr.When i checked the records i found that rtt was always zero.To double check i turned on rtcp debug I was only able to see sent rtcp,sending rtcp but no got rtcp. Also the timestamps in rtp debug were oddly dissimilar.Is there a prblem wth my timestamp or is this a known bug? --...

...229.18:37910 has been created [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:27143 has been deleted [2016-03-21 20:39:58] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:37910 is now Reachable. RTT: 41.882 msec [2016-03-21 20:41:01] VERBOSE[28019] res_pjsip/pjsip_configuration.c: Contact 17367/sip:17367 at 46.39.229.18:37910 is now Unreachable. RTT: 0.000 msec [2016-03-21 20:41:06] VERBOSE[3827] res_pjsip_registrar.c: Added contact 'sip:17367 at 46.39.229.18:60105' to AOR '17...

...on Apple Xserve i7 server with PCI cards for 4GB FC (LSI FC949ES) and 10GB Ethernet (Intel 82598EB) dom1: Centos 5 OS with passthru mode of PCI 4GB FC and 10GB Ethernet (ixgbe and mptfc blacklisted in dom0) Everything starts ok, but networking in dom1 using the 10GB interface has extremely high rtt w/ping tests (20-30ms avg). I've tried a few tweaks I've discovered via googling, but nothing seems to help. (run libvirt as root, set privilege mode, etc) For validation, I have the same card in Dell hosts running Centos 6 natively and they get an avg rtt of .06ms or better. Furthemore t...

I've upgraded one of my client boxes to Windows 64 bit edition but now it cannot login to the Samba PDC, instead it gives a RPC error after entering the password. The Win32 boxes can still login fine. I can get accounts without admin rights to login but those with the rights are given a message stating "The system cannot log you on due to the following error: A remote procedure call

I have a customer who is complaining about slow SFTP transfers over a long haul connection. The current transfer rate is limited by the TCP window size and the RTT. I looked at HPN-SSH, but that won't work because we don't control what software the peer is using. I was thinking about coding a much more modest enhancement that just does SO_RCVBUF for specific subsystems. In the interest of adding something that the OpenSSH community would take back...

I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1<called-num>@voice.google.com,,rTt") in new stack Otherwise all works. The call goes through, good audio. sean

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

...eral aors in the table ps_aors, like this (real url replaced by myurl): *CLI> pjsip show aor pbx-node-1 Aor: <Aor..............................................> <MaxContact> Contact: <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)..> ========================================================================================= Aor: pbx-node-1 0 Contact: pbx-node-1/sip:myurl:5060 771bf6a7d4 Created 0.000 ParameterName : ParameterValue ============...