I picked up a Chateau LTE12 which arrived with RouterOS v7.0beta6 factory installed. For initial testing no changes were made to the default configuration other than to set passwords for the admin user and WiFi access.

With v7.0beta6, I had a basic VoIP system physically connected to ether5 which successfully registered with it's service provider and was able to make and receive calls.

Upgraded to the latest (RouterOS v7.0beta8) and the VoIP system failed to register with it's service provider, downgraded to RouterOS v7.0beta7 with the same results.

I have submitted a support request to have RouterOS v7.0beta6 as this version does not appear to be available for download. In the interim, I have downgraded to RouterOS v7.0beta5 and VoIP is able to register with it's service provider, make and receive calls again. I was only able to downgrade RouterOS, Routerboard would not downgrade, presummably because beta5 is a lower version than the factory installed beta6.

The VoIP system is a Gigaset SL450A (with Go-Box 100), VoIP service provider is Sipgate, net access is via the Chateau's LTE interface with no public IP address on my side.

Other than to say "Works with this version but not with that version", is there anything else I can do at my end to provide Mikrotik with more detail to resolve this issue (enable extneded logging or debug options etc,.)?

you should create a supout.rif at both situation and send those files to MikroTik Support Team, only thay can fix it.

Remember that SIP Helpers should be disable in Firewall\Service.

Thank you, I will look to create the supout.rif file for both situations.

I did already check the SIP helpers option, on both working and non-working versions this is enabled by default, on the non-working versions I had tried disabling the option but this had no affect.

I also tried to see if any option or parameter values had changed between upgrades but I could not see any, although, I may have missed something.

...I may have missed something.

Tool Sniffer + https://www.youtube.com/results?search_ ... shark+voip

I spent some time on this earlier today. I'm not seeing much difference between the packet captures on beta5 (working) version vs beta8 (non-working).

On both versions, the VoIP systems goes through the same process and sends out SIP register requests. On beta5, after sending the SIP register requests, the VoIP system occasionally sends UDP packets (src/dst port 5060) to the SIP server, these are the NAT keep alive packets. On beta8, after the SIP register requests, there is one UDP keep alive packet sent, then it goes back to sending register requests and eventually restarts the whole process.

I added firewall to the logging to see if there was anything that may be getting blocked or dropped but nothing from the firewall appeared in the logs.

On both versions I don't see any SIP registration confirmation/success response packets, but as I'm unaware of the finer details of the SIP protocol, I'm not sure if confirmation is expected. From looking at the packet capture of the working version in Wireshark I can't see how the VoIP system knows if it's successfully registered or not.

I spent some time on this earlier today.

All my analyze with WireShark and VoIP are like that:
Menu Telephony > VoIP Calls
I select the call and open "Flow Sequence" and do screenshot.

I do the same with other one and compare both screenshot's - the difference like, differ ports, no RTP streems.... give me information about communication.

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Menu Telephony > VoIP Calls > Play Streams give me feedback about problems like quality etc.
I grab/sniff cap file from last branch router who have end user and from DC router who have a VoIP PBX - with that I can play recorded sound who is near PBX and Phone and listen problems at both directions.

Maybe this help you.

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Remember that some PBX need have got DNAT and ros in beta 5|6 vs 7|8 maybe not give you Public IP and that's why this not work. Then this will be outside wireshark with voip filter.

With beta7/8 I don't get as far as being able to make/receive calls, the issue is that the VoIP system fails to register with the VoIP service provider (or I think it might register briefly but then permanently drops the registration).

As mentioned in my initial post, there is no public IP on my end. The net connection is via the LTE interface and the cellular provider does not provide a public IP address.

However, the lack of public IP address is the same regardless of RouterOS version, the VoIP system on beta5/6 is fine without public IP address.

The only variable is the RouterOS version, upgrading or downgrading, the configuration appears unchanged as far as I can tell. So at this time, I can only conclude that my VoIP system is being handled differently in beta5/6 than it is in beta7/8. Unfortunately, it's the later software versions that render my VoIP system unusable and consequently, it renders the Chateau LTE12 router unusable for me.

With beta7/8 I don't get as far as being able to make/receive calls, the issue is that the VoIP system fails to register with the VoIP service provider (or I think it might register briefly but then permanently drops the registration).

This will be show at wireshark and you should compare that both systems.
If you want you could share 2 cap file via telegram in my signature and I can give you my feedback to it. At cap I will be see numbers and ip addresses.

After going back and paying a little more attention while looking at the pcap files, I can see that only the packets in one direction are being captured, both beta5 and beta8 are only capturing the TX packets from the VoIP system (packet sniffer direction is set to "any"), I guess that's another bug/issue.

I also have another MikroTik unit here (LtAP LTE6) running RouterOS 6.47, the VoIP system works fine with that and using the packet sniffer tool does capture packets in both directions and I can now see the registration request confirmation responses, so I now have a complete capture and can see the expected process flow.

I did notice another issue with RouterOS v7, when downloading the pcap files, the filename offered up appears to be random and ends with a .dms file extension. Even after changing the filename and extension in the "Save As" dialog the downloaded file is saved with a .dms extension, when downloading the pcap file from RouterOS v6.47, the are offered up with the correct filename and extension.

It looks like I'll have to stick a switch inline between the VoIP system and the Chateau and then mirror the switch port out to Wireshark in order to get a complete packet capture.

After taking a little bit of a crash course in all things SIP and learning way more than I ever wanted to about SIP ALG, I think I have taken this as far as I can.

Sticking the packet sniffer on to the WAN (lte1) interface will capture packets in both directions, in beta5 the VoIP system sends out SIP register requests to the server and receives confirmation/success responses. In beta8 the same SIP register requests are sent out but no reply is ever received, it is unclear if the server never receives the requests from the router or if the router never receives the response from the server. As this is an external server, I cannot take this any further.

What I know or have confirmed thus far is that the VoIP system works with MikroTik LtAP LTE6 running RouterOS v6.47 and also works with MikroTik Chateau LTE12 running RouterOS v7.0beta5/6 and *does not* work with MikroTik Chateau LTE12 running RouterOS v7.0beta7/8.

During all this I also noticed that SIP ALG/helpers in (at least) RouterOS v7.0beta5/8 just don't work, enabled or disabled the SIP messages still contain the internal LAN IP address, with the LtAP LTE6 RouterOS v6.47 with SIP helpers enabled the relevant IP addresses within the SIP messages are re-written with the IP address of the WAN interface.

Sticking the packet sniffer on to the WAN (lte1) interface will capture packets in both directions, in beta5 the VoIP system sends out SIP register requests to the server and receives confirmation/success responses. In beta8 the same SIP register requests are sent out but no reply is ever received, it is unclear if the server never receives the requests from the router or if the router never receives the response from the server. As this is an external server, I cannot take this any further.

That two files you should report to MikroTik Support - really, they can fix this.

I have generated supout.rif and packet capture files when running on both beta5 and beta8 and submitted those to support. Hopefully they will accept this as an actual issue and that the details within files provide enough information to diagnose and address the issue.

Hello, is there any solution available for VoIP on v7.0beta8 ?
Best regards

Hi Redmonkey,

Any news about your SIP issue ? I have the same problem with chateau v7, no problem with another devices with the lastest release of v6.

redmonkey

and submitted those to support.

What support say ?
If new Ros7.1beta fix this issue ?

Hey Redmonkey,

Thanks for highlighting this issue.
I've experienced the same thing with a Yealink T53W handset on a Chateau running 7.1beta2
I reverted back to my HAP ac2, running 6.47.7, no issues until I upgrade it to 7.1beta2 (fortunately with the HAP you can downgrade, back to version 6, unlike the Chateau).
I ran a Packet sniffer/Wireshark from the router, and I can see the phone sending the packets to the router, but they don't pass through to reach the SIP server.
I have disabled the SIP Helper, but makes no difference.

I'm waiting for the next development release, which hopefully resolves this issue.

However Acrobits (softphone) on my mobile phone logs into the same SIP server without an issue using 7.1beta2

Kind regards,
Ryan van Klaveren

I've experienced the same thing with a Yealink T53W handset on a Chateau running 7.1beta2
I reverted back to my HAP ac2, running 6.47.7, no issues until I upgrade it to 7.1beta2 (fortunately with the HAP you can downgrade, back to version 6, unlike the Chateau).
I ran a Packet sniffer/Wireshark from the router, and I can see the phone sending the packets to the router, but they don't pass through to reach the SIP server.
I have disabled the SIP Helper, but makes no difference.

However Acrobits (softphone) on my mobile phone logs into the same SIP server without an issue using 7.1beta2

Still experiencing the same issue after upgrading to 7.1 beta3

Please create acount at https://mikrotik.com/support , best way using the same e-mail what was before then you receive the same list of cases reports prev as support@mikrotik.com .
Then this case with VoIP please share with me as info@sibex.it

I help speak with support to solve that case

This issue was fixed in 7.1beta3 and all those who wrote to support, reported that the issue is gone now. Make sure you upgraded to the latest version from our download page. You can also try to reset the configuration and reconfigure the unit.

It's FINALLY working!!!!! THANKS SO MUCH!! I've been struggling with this problem for DAYS and now it's perfectly working!!!! Love u guys from mikrotik team, THANKS