VoIP calls dropping
Hello All I am having an issue with a client who i am doing VoIP. Calls can be ongoing then they suddenly drop.Doing a simple traceroute i am able to reach the client's server and router. Ping tests indicate a round trip of about 20-30ms.I dont think this can affect the quality.What cud be the cause of the call drops? Also is there a tool i can use to measure VoIP perfomance. Also note that i have not done any QoS because the calls are too few Thanks ---------------------------------------------- This message has been scanned for viruses and dangerous content by Jambo MailScanner, and is believed to be clean. --------------------------------------------- "easy access to the world"
Hi Cynthia, What codec are you using for the calls? Also do you have any idea what kind of queueing is in use on your/your customers edge routers? Regards, Brian On Thu, Nov 5, 2009 at 9:40 AM, Cynthia Wahome <cwahome@jambo.co.ke> wrote:
Hello All I am having an issue with a client who i am doing VoIP. Calls can be ongoing then they suddenly drop.Doing a simple traceroute i am able to reach the client's server and router.
Ping tests indicate a round trip of about 20-30ms.I dont think this can affect the quality.What cud be the cause of the call drops? Also is there a tool i can use to measure VoIP perfomance. Also note that i have not done any QoS because the calls are too few
Thanks
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am actually not a VoIP expert.i have just inherited this setup from someone who left. Here are the codecs G. 711 A Packet size 20ms G. 711 MU Packet size 20ms G. 729 Packet size 20ms G. 729AB Packet size 20ms G.729B Packet size 20ms
Hi Cynthia,
What codec are you using for the calls? Also do you have any idea what kind of queueing is in use on your/your customers edge routers?
Regards,
Brian
On Thu, Nov 5, 2009 at 9:40 AM, Cynthia Wahome <cwahome@jambo.co.ke> wrote:
Hello All I am having an issue with a client who i am doing VoIP. Calls can be ongoing then they suddenly drop.Doing a simple traceroute i am able to reach the client's server and router.
Ping tests indicate a round trip of about 20-30ms.I dont think this can affect the quality.What cud be the cause of the call drops? Also is there a tool i can use to measure VoIP perfomance. Also note that i have not done any QoS because the calls are too few
Thanks
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-- Brian Munyao Longwe e-mail: blongwe@gmail.com cell: + 254 722 518 744 blog : http://zinjlog.blogspot.com meta-blog: http://mashilingi.blogspot.com
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Cynthia, Cynthia Wahome wrote:
am actually not a VoIP expert.i have just inherited this setup from someone who left. Here are the codecs G. 711 A Packet size 20ms G. 711 MU Packet size 20ms G. 729 Packet size 20ms G. 729AB Packet size 20ms G.729B Packet size 20ms
This are the ones listed - but which one is the default in use?. G.711 is high on bandwidth utilization. G.729 is your best bet but its proprietary - meaning that not all IP phones have it on by default- this is something you may want to check. Set the default codec to be the one all your devices can support. Often enough most will support G.711. HTH. Regards, Mich
Hi Cynthia, Cynthia Wahome wrote:
Ping tests indicate a round trip of about 20-30ms.
Ping tests will establish if theres end-to-end reachability I dont think this can
affect the quality.
Right.
What cud be the cause of the call drops?
There could be a variety of problems. Please enable logging of the Server side and monitor to see the cause of call termination.
Also is there a tool i can use to measure VoIP perfomance.
VOIP is sensitive to jitter. One of the causes of jitter can be packet-loss. You may want to conduct a couple of tests (end-to-end) tests using something like iperf to see what throughputs are available. Secondly monitoring the link using smokeping or something like mtr would tell you where the packet-losses are.
Also note that i have not done any QoS because the calls are too few
Whats the current link utilization like? Regards, Michuki.
Thanks Michuki The link utiliziation is only at 19%. I will do as suggested and give feedback. Thanx
Hi Cynthia,
Cynthia Wahome wrote:
Ping tests indicate a round trip of about 20-30ms.
Ping tests will establish if theres end-to-end reachability
I dont think this can
affect the quality.
Right.
What cud be the cause of the call drops?
There could be a variety of problems. Please enable logging of the Server side and monitor to see the cause of call termination.
Also is there a tool i can use to measure VoIP perfomance.
VOIP is sensitive to jitter. One of the causes of jitter can be packet-loss. You may want to conduct a couple of tests (end-to-end) tests using something like iperf to see what throughputs are available.
Secondly monitoring the link using smokeping or something like mtr would tell you where the packet-losses are.
Also note that i have not done any QoS because the calls are too few
Whats the current link utilization like?
Regards,
Michuki. _______________________________________________ Skunkworks mailing list Skunkworks@lists.my.co.ke http://lists.my.co.ke/cgi-bin/mailman/listinfo/skunkworks ------------ Skunkworks Rules http://my.co.ke/phpbb/viewtopic.php?f=24&t=94 ------------ Other services @ http://my.co.ke Other lists ------------- Announce: http://lists.my.co.ke/cgi-bin/mailman/listinfo/skunkworks-announce Science: http://lists.my.co.ke/cgi-bin/mailman/listinfo/science kazi: http://lists.my.co.ke/cgi-bin/mailman/admin/kazi/general
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@Cyn..th...ia............call drop, engaged tone........ :-))) As Mich has also pointed out, packet losses could be the main cause. To get away from QOS, jitter and the rest try to get a separate circuit for voip which will eliminate the issues, but first do check the for link drops and packet losses to establish cct parameters. Using QOS and the rest will affect quality at some point eg Monday morning when mail download traffic is high. HTHs.
@Cynthia, I have following this thread. I just had a thought. QoS is recommended for real-time applications. One of the mechanisms is header compression and packet fragmentation (cRTP and LFI). The reason as to why this is done is because VoIP packets are smaller compared to say FTP packets. If FTP and VoIP packets are sent using the same link, the serialisation delay for the FTP packets could be so significant such that the VoIP packets are delayed or droppped. Please remember that the latency from interactive applications such telnet may differ from the real-time latency since the serialisation delay for telnet packets is quite low. Just to paint the picture, the avearge VoIP packet is 40 bytes(IP,UDP,RTP) while the FTP packets could be 150 bytes (IP,TCP,Payload). My recommendation, troubleshoot layer 2 and 3. If it doesn't work, try QoS first. @Michuki, Thanks for info. I have learnt a lot. On Thu, Nov 5, 2009 at 11:06 AM, aki <aki275@googlemail.com> wrote:
@Cyn..th...ia............call drop, engaged tone........ :-)))
As Mich has also pointed out, packet losses could be the main cause. To get away from QOS, jitter and the rest try to get a separate circuit for voip which will eliminate the issues, but first do check the for link drops and packet losses to establish cct parameters. Using QOS and the rest will affect quality at some point eg Monday morning when mail download traffic is high. HTHs.
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participants (5)
-
aki -
Brian Munyao Longwe -
Cynthia Wahome -
Michuki Mwangi -
techi