Other party cannot hear me starting within the first 1-3 minutes of the call

[Peter4487]

I am now experiencing the issue first reported in BelledonneCommunications/linphone-android#1848 very intensely with voip.ms with the last update to 5.0.11. Before it would happen within the first maybe 20 minutes, now it's within the first couple minutes.

The mic stops working (or at least the other person can't hear me) within the first few minutes of the call. Logs attached, I sent the log shortly after the mic stopped working.

This was with linphone-android 5.0.11 (tags/5.0.11^0, release) & Linphone SDK 5.2.58 (tags/5.2.58^0, release), voip.ms, Pixel 5 running GrapheneOS.

linphone_logs_cleaned.txt

Originally posted by @Peter4487 in https://github.com/BelledonneCommunications/linphone-android/issues/1848#issuecomment-1557520368

[Viish]

Have you tried without the VPN to rule that out? From your logs everything seems fine, the issue may be network related or due to your correspondent device/softphone. Can you get its logs?

[Peter4487]

Thanks for looking in to this @Viish !

I have tried it with and without the VPN. The problem is actually worse without the VPN. Mic cuts out and then call fails within 30 seconds.

Concerning the correspondent device, I am just calling a desk phone somewhere in my building from my device (android cell phone). I assume it's some sort of hardware softphone since it has CAT-5 going in to it (It's a Polycom something or other).

I've attached logs of the no-VPN call. Mic stopped working about 22 seconds in. I've repeated this test several times.

linphone_logs_no_vpn_cleaned.txt

[Viish]

Thansk for the new logs.

Can you try to call after disabling SRTP as media encryption in the settings? Also, would it be possible for you to use another audio codec?

[Peter4487]

I disabled the Opus codec and still see the same problem (disconnect in ~30s with no VPN).

With the VPN but SRTP disabled everything seemed to work for a single call after changing the settings (in linphone-android and on voip.ms). After that, we're back to the mic doesn't work after about 30 seconds with TCP or UDS. I don't think whether the VPN was on or not made a difference. Attached are some logs.

If there's something else you'd like to see please let me know.

linphone_logs_no_SRTP_cleaned.txt

[Peter4487]

Update: voip.ms works fine with baresip android and SRTP encryption in case that's informative.

[2dengine]

I am experiencing a similar issue where the correspondent's voice would be drowned by static (they can hear me but I cannot hear them). This issue only occurs with the TCP/UDP protocol and does NOT occur when using other SIP clients. I noticed that my version of Linphone is missing the g729 codec from the settings screen. Any tips would be greatly appreciated.

[UPBT]

I disabled the Opus codec and still see the same problem

I don't think voip.ms supports Opus.

https://wiki.voip.ms/article/Codecs_Supported

G.711u, G.729a, GSM

also G.722 (BETA not in wiki yet)

[UPBT]

I was recently getting a run with this same issue within 1 minute of dialed calls. I am also using voip.ms as the provider.

I get this when not using a headset, I also experienced this issue after 5-10 minutes with an earlier version of Linphone but I was using a headset. At the time I blamed the headset.

To try to diagnose the issue, I called the voip.ms echo test line (4443) and stayed on the line for about an hour. I did not experience any issues in that hour.

The problem seems highly intermittent.

[alex8065]

This is happening for me as well, same provider (voip.ms), Pixel 5 also running GrapheneOS.

@UPBT, for whatever reason, this doesn't happen on 4443 for me either.

[Peter4487]

voip.ms says others have reported problems with linphone to them, and they think the problem is with linphone. I don't have a problem with voip.ms + baresip, but that combo eats a lot of battery.

[UPBT]

@UPBT, for whatever reason, this doesn't happen on 4443 for me either.

I've had it happen on other outgoing calls, but most of the time it appeared when I called a virtual receptionist (a/k/a voicemail hell) used DTMF to navigate a menu, and was transferred to a real human. I got halfway through with the person and all of a sudden they couldn't here me but I could still hear them.

I did make some outgoing calls with the same issue popping up 30 seconds to 2 minutes into the call, talking to a human the whole time and not being transferred.

Unfortunately, I "fixed" the issue by having to resort to dialing out on my provider's line (conventional cellphone, although it's probably got VoLTE going on and is thoroughly debugged.

Never an issue with dialing 4443 (voip.ms echo test) but that call presumably went to their servers and stayed there.

Unfortunately I don't know how to test this without needlessly harassing real working humans at the other end of toll-free phone numbers. I'm open to suggestions. I want this bug squashed.

[sk425]

I have the same issue using SIP Provider: voip.ms App version: linphone 5.1.4 f-droid arm64-v8a OS: grapheneos Phone: pixel 6

However I didn't follow the troubleshooting guide here https://wiki.voip.ms/article/Call_quality_issues#One-Way_Audio

Excerpt:

• [ ] From the account or sub account settings, select always NAT=Yes (this is the option recommended by VoIP.ms).
• [ ] Try using each codec in a separate way, starting with G711u codec only, from the customer portal > Main menu > Account settings > Advanced tab > allowed codecs.
• [ ] Only as a test, place the device in DMZ to test if the issue is related with the NAT. do not leave the device in DMZ after finishing troubleshooting.
• [ ] Is your router appropriate for VoIP? If you have a router and a modem, try to bypass the router to verify if the issue gets duplicated.
• [ ] If your router includes a SIP ALG and/or SPI Firewall setting please ensure that it is disabled. That setting is common in D-Link and Netgear routers. If this does not help make sure you are using the most recent firmware version for your device.
• [ ] You can see what port is being used for audio by looking at the UDP port 5060 traffic. The RTP traffic will typically be in the UDP port range 10000-20000.

I realised two things:

1. I selected all codecs on both sides though linphone has many more than voip.ms. I then narrowed the selection on linphone to only the compatible ones.
2. The port range was random on linphone side. I selected 42873 as the third compatible encrypted option (as an obscure option in case default sip ports are blocked on a network i'm currently on, idk lol)

Will see how things go. If I don't post back for a while, I guess we can assume it went well.