Low incoming audio on SIP trunks

Asterisk version 13.15.0
FPBX version
SIP trunks

This is a new one on me. Intermittantly, about 15% of the calls the system handles has an inconsistant incoming audio level. What I mean is that the callers voice will start out just fine and as the call progresses the caller volume level will decrease to a point where the call will drop. Sometimes the volume level will return to normal. The SIP trunks are being sent out over a Fiber connection. Chan-sip info shows the connect time <30ms. This audio problem is also exhibited when a caller leaves a voicemail message as well, so we do not suspect the phones. We have TCPdump running but have not captured a bad call yet. SIP trunk provider issue? Something going on with the Fiber? As I said this is new to me. Not one complaint from the outside that they cannot hear the user.



if the problem does not occur using internal phones, then my guess is that you have a bandwidth management issue on your internet connection. you need to make sure that you have enough bandwidth to handle the call. if your router is not doing bandwidth management it should. i don’t know how your fiber is connected. if it is a dedicated fiber connection it will be a more consistent (bandwidth wise) than a shared connection (i.e. comcast coax). bandwidth management is used to ensure you always have sufficient bandwidth in both directions for calls. without it, simply the act of uploading or downloading a large file could impact voice quality.

We have a separate network setup from the data guys. There is however a single Wan port that is shared with them. the fiber Modem has a single network output. We plugged that into a Gig 5 port switch and then our router and their sonicwall into it. We have separate Wan IPs. the switch is managed (Netgear). We prioritized our traffic over the data by port. I do not know what the Fiber speed or contract speed is at this time.

it is exactly this setup that you want to avoid - two separate networks using the same internet connection without bandwidth management. there are probably a number of ways to solve this problem, but the one we usually come back to is the Edgemarc 4700 or the new 2900 session border controllers. these allow you to do bandwidth management on the wan interface for everything and still allow your IT guys to have their separate network.

Well one part of this saga is probably solved. I caught a phone sending a Bye .2 sec after it established an RTP stream for a call. Going to try and update the firmware for that. The other is still out there. Please correct me if I am wrong, but isn’t it almost impossible to “adjust” audio levels in a SIP call? Obviously the user can turn the audio up and down on their phone as the call progresses, but in these cases the audio fades to about 1/2 the level it started out and sometimes it comes back to normal. An SBC wouldn’t help in either of these instances would it? We have discussed installing SBC’s in the office for customers, but to no conclusion.