I would like to comment just on the download QUE tree section.. We make use of a Cloudcore so instead of the specifiying the LAN bridge as per your example for traffic originating from your WAN connection we have to select the Eth 1 (LAN Interface) My question is the following... Wouldnt this limit all traffic speed from other interfaces (branch office WAN for example) to the Eth 1 LAN interface ?? For example im my case just for testing Ethernet 1 is where all the phones would most likely be connected. Wouldnt specifying ethernet 1 limit all traffic from other interrfaces to ethernet 1 ???? Or would a better idea be to put all VOIP phones on a VLAN and specifiy the VOIP vlan as the exit interface in the Download Parent Section ? Thanks
Thank you very much! Very good presentation, highly useful . Only good stuff. Also I like to see Linux Mint OS used on presentation laptop. It is what I am using on my working laptop too.
Thank you, and yes I love linux mint! however I started using dropbox in place of google drive because drive (grive) is not reliable in linux and not supported by google??!!!
Absolutely agree. I used Gdrive and after Onedrive. Onedrive updates just crash when I do not have time to look why is crashed. I was saw dropbox supported by Ubuntu. So it will be my next step for personal documentation storage.
Hello, callers cannot hear me when they call in (I can hear them), and on another VoIP app, the calls are oftentimes dropping. I was told that SIP ALG needs to be disabled, however, I work from on a phone data plan that I tether to my laptop. Will the phone company be able to disable SIP ALG in this regard, or what other options may I look into? Thank you!
I have several customers using OOMA Voip and OOMA doesn't manage the SiP the same way you describe in the video where I can specify the SiP Destination IP. From OOMA support, "Unlike most other phone providers, we securely tunnel all SIP communications to our ooma SIP Servers." I am not sure how to mark the connection without the SiP IP. BTW: I have about 6 customers using OOMA and they are all experiencing choppy outgoing call quality (heard from their connected customer but not heard by them.) I want to set up queues but am stuck at the mark connection step. Any Tips? Awesome video BTW. Learned a lot.
@@yeo1397 The only time you should specify egress port (because you can only queue packets leaving an interface) is if you can not specify the traffic by other means. In the case of SIP and RTP packets, there are very well defined ways to specify those traffic types.
@@SkibidiWaPaPaPaPaThanks for the reply! Also, I have another question. I know that this is an irrelevant question in your presentation, but this is bugging me for a long time. When I use PCQ, does the limit is being measured by Packets or KiB in the latest RoS version? I was very confused about this parameter. Queue Size in KiB, but how do I convert KiB into a number of packets just like the default fifo queue type?
hey i really need some help i am doin dual wan load balancing with mikrotik and my clients whatsapp calls are not connecting nor the voip box dont register? can you help me what exactly i should do.. please and thank you
I'm not even sure that whatsapp uses SIP protocols. I would start with finding out what protocols whatsapp uses OR find out how whatsapp server subnets are, that way you can prioritize by subnet / ip destination rather than protocol. If it is using SIP protocols please reference my video on SIP ALG as thats where I talk about SIP problems, one of which the problem you are describing. ua-cam.com/video/tM7wyKdnIKA/v-deo.html
I use 2 isp, and use 2 Mikrotik with different segments, segment 192.168.216.0/24 on Mikrotik 1 isp 1, and segment 192.168.223.0/24 on Mikrotik 2 isp 2, but when one isp has a problem, for example ISP 1 down, I move the segment path 192.168.216.0/24 through mikrotik 2, with this configuration there is one way call, can you tell me how to fix it ??
Thank you for your effort a really infomative lesson thank you for sharing.
I would like to comment just on the download QUE tree section.. We make use of a Cloudcore so instead of the specifiying the LAN bridge as per your example for traffic originating from your WAN connection we have to select the Eth 1 (LAN Interface)
My question is the following... Wouldnt this limit all traffic speed from other interfaces (branch office WAN for example) to the Eth 1 LAN interface ??
For example im my case just for testing Ethernet 1 is where all the phones would most likely be connected.
Wouldnt specifying ethernet 1 limit all traffic from other interrfaces to ethernet 1 ????
Or would a better idea be to put all VOIP phones on a VLAN and specifiy the VOIP vlan as the exit interface in the Download Parent Section ? Thanks
@22:38 you have command line at priority 8 which is supposed to be 1 for the parent...but I guess the parent priority is not used anyways.
You just made my day. Thank you
Thank you very much! Very good presentation, highly useful . Only good stuff. Also I like to see Linux Mint OS used on presentation laptop. It is what I am using on my working laptop too.
Thank you, and yes I love linux mint! however I started using dropbox in place of google drive because drive (grive) is not reliable in linux and not supported by google??!!!
Absolutely agree. I used Gdrive and after Onedrive. Onedrive updates just crash when I do not have time to look why is crashed. I was saw dropbox supported by Ubuntu. So it will be my next step for personal documentation storage.
Hello, callers cannot hear me when they call in (I can hear them), and on another VoIP app, the calls are oftentimes dropping. I was told that SIP ALG needs to be disabled, however, I work from on a phone data plan that I tether to my laptop. Will the phone company be able to disable SIP ALG in this regard, or what other options may I look into? Thank you!
Great Job, Thank You!
Thank you, I can walk by any phone and it echoes when I walk by the phone.
1. one way traffic call dialed is ok if any one wants to call its gives the phone busy
2. one way audio also is one more problem
could be SIP ALG
I have several customers using OOMA Voip and OOMA doesn't manage the SiP the same way you describe in the video where I can specify the SiP Destination IP. From OOMA support, "Unlike most other phone providers, we securely tunnel all SIP communications to our ooma SIP Servers." I am not sure how to mark the connection without the SiP IP. BTW: I have about 6 customers using OOMA and they are all experiencing choppy outgoing call quality (heard from their connected customer but not heard by them.) I want to set up queues but am stuck at the mark connection step. Any Tips? Awesome video BTW. Learned a lot.
Start with setting your customers router with prioritization rules for their upload bandwidth.
What was the program you said you used to test customers or potential customers networks with?
Did you get the name?
Great information. Thanks for sharing!
thank you, if voip is of interest to you, I also have a video on SIP ALG ua-cam.com/video/tM7wyKdnIKA/v-deo.html
@@SkibidiWaPaPaPaPa Is it best to specify interface when marking the packets with destination ports? or should be leave out blank for cpu efficency?
@@yeo1397 The only time you should specify egress port (because you can only queue packets leaving an interface) is if you can not specify the traffic by other means. In the case of SIP and RTP packets, there are very well defined ways to specify those traffic types.
@@SkibidiWaPaPaPaPaThanks for the reply! Also, I have another question. I know that this is an irrelevant question in your presentation, but this is bugging me for a long time. When I use PCQ, does the limit is being measured by Packets or KiB in the latest RoS version? I was very confused about this parameter. Queue Size in KiB, but how do I convert KiB into a number of packets just like the default fifo queue type?
@@yeo1397 wiki.mikrotik.com/wiki/Manual:Queues_-_PCQ look at the section "PCQ rate examples". What you are looking for is there.
hey i really need some help i am doin dual wan load balancing with mikrotik and my clients whatsapp calls are not connecting nor the voip box dont register? can you help me what exactly i should do.. please and thank you
I'm not even sure that whatsapp uses SIP protocols. I would start with finding out what protocols whatsapp uses OR find out how whatsapp server subnets are, that way you can prioritize by subnet / ip destination rather than protocol. If it is using SIP protocols please reference my video on SIP ALG as thats where I talk about SIP problems, one of which the problem you are describing. ua-cam.com/video/tM7wyKdnIKA/v-deo.html
Good information... I would love to work for your co. :
Pretty Simple, pretty nice!!!
Wireshark and proper VoIP/Network Engineering concepts = Mangle.
Hi, I would like to slide this class, thank you.
It's on the Mikrotik website. Look at past conventions, the PDF is linked on the video
sir my hosts page is empty but ip binding page is full i dont now why my hosts page is empty sir plz give me solution
I use 2 isp, and use 2 Mikrotik with different segments,
segment 192.168.216.0/24 on Mikrotik 1 isp 1, and segment 192.168.223.0/24 on Mikrotik 2 isp 2, but when
one isp has a problem, for example ISP 1 down, I move the segment path 192.168.216.0/24 through
mikrotik 2, with this configuration there is one way call, can you tell me how to fix it ??
Hi David, what is you personal email? I would like to communicate with you. regards.
indonesia
He's so hot.