Does Sip Helper app (SIP ALG) identify audio packet streams set up so they will be marked so they can be prioritized or guaranteed a chunk of bandwidth in a simple Queue? Or even to route SIP out a second ISP connection.
I want to automatically recognise VOIP audio streams that the SIP headers define and be able to have them be part of a VOIP queue that is guaranteed it’s bandwidth.
I understand setting up the queue would not be automatic, I’m just hoping the audio packets will be marked automatically. So basically what I’m hoping the SIP Helper is doing as a call is started and ended is that Sip Helper sets up and tears down firewall mangle rules that are based on the SDP that the SIP Helper already parsed as it “mangles” it for nat.
What I would really like to see in SIP Helper is the option to not mangle the SIP headers and the option to mark audio stream packets to be managed by queues.
But I don’t find any info on any connection between SIP Helper and Queues.
It seems the only info on using queues for voip involves individually identifying the IPs of VOIP phones and marking them. https://wiki.mikrotik.com/wiki/Voip
That solution is not dynamic and scalable or support softphones that are on a computer that has mixed internet traffic.
I work for a multi tenant hosting provider and am looking for this functionality so I can recommend Mikrotik routers to our resellers as the best way to ensure last mile audio quality. As most know, last mile congestion is almost always the source of bad audio and a bad SIP ALG the cause of broken SIP. From the sound of it Mikrotik has done SIP ALG right but haven’t been able to find if it assists with the audio like I described above.