Call drops every 4min

[shopereira]

Hi guys,

I'm using freephoneline (Montreal number) and every income call I get it drops after 4min30seg. I'm using the desktop app version (windows 10). For the 4min of the call. .. it is just perfect..very good quality.
any advice would be much appreciated.

thx,

[Liptonbrisk]

1) Ensure whatever modem/router combo (or gateway) your ISP gave you is in bridge mode if you are using your own router.


2) Do all the calls drop exactly at 4:30?
If so, in your router, try disabling SIP ALG.
See here for a generic example: https://www.obitalk.com/info/faq/sip-alg/disable-alg.
If your router was issued by your ISP, contact your ISP. In some routers, the feature is hidden with no way for consumers to disable it (sometimes only the ISP can)

To see what problems can be caused by buggy SIP features in routers, visit https://www.voip-info.org/routers-sip-alg/ (scroll down to SIP ALG problems).

3) If the calls are dropping randomly, properly enable QoS in your router for your computer that's running the Freephoneline desktop app (and ensure no other programs are running on your computer that are hogging bandwidth while using the Freephoneline desktop app). Refer to your router's manual or contact your ISP if you were issued a modem/router combo from them (typically those routers have horrible or absent QoS functions).
For a general explanation of QoS, visit: http://www.voipmechanic.com/qos-for-voip.htm
Avoid anything it says about the G.729 codec.

a) Also visit https://forums.redflagdeals.com/newegg- ... #p28508733. Bad jitter/huge ping spikes can cause calls to drop.

4) Visit https://forums.redflagdeals.com/freepho ... #p29299104 to see what features are important to have in routers for use with SIP services

[shopereira]

Hi Liptonbrisk,

1) I'm running it under my corporate network. I already checked in the firewall all the ports to be open for this purpose.
2) I passed a wrong (incomplete) info (sorry). The call doesn't drop actually. The other side person stop listing myself (I can hear he/she but he/she cant listen at all). That is for incoming call so far, I haven't tested outgoing call just yet. But yes, I loose my audio every 4:30

thx,

[Liptonbrisk]

Hi Liptonbrisk,

1) I'm running it under my corporate network. I already checked in the firewall all the ports to be open for this purpose.

SIP ALG problems aren't necessarily port issues. While this may not be SIP ALG related, I would still try disabling it, for testing purposes, if it's enabled.

2) I passed a wrong (incomplete) info (sorry). The call doesn't drop actually. The other side person stop listing myself (I can hear he/she but he/she cant listen at all).

Outgoing audio is affected by available upload (from you to the server) bandwidth.
In that case, what I wrote previously applies:

Properly enable QoS in your router for your computer that's running the Freephoneline desktop app (and ensure no other programs are running on your computer that are hogging bandwidth while using the Freephoneline desktop app). Refer to your router's manual or contact your ISP if you were issued a modem/router combo from them (typically those routers have horrible or absent QoS functions).
For a general explanation of QoS, visit: http://www.voipmechanic.com/qos-for-voip.htm
Avoid anything it says about the G.729 codec.
Also visit https://forums.redflagdeals.com/newegg- ... #p28508733. Bad jitter can cause audio to break up.

If the problem isn't bandwidth/jitter (ping spike) related, then unfortunately, without a SIP trace or packet inspection, I can't comment further, except that it's probably NAT related. I don't know whether the Freephoneline desktop app attempts to send a reinvite/session refresh around the 4:30 mark (seems unlikely to me, but I don't get paid enough to test). If so it's possible that one port is used for outbound audio, but after a reinvite, the session refreshes using a different port and, in turn, effectively kills outbound audio for you due to a NAT related issue. I would also begin to consider a UDP timeout problem, but I'm guessing. Regardless, it is useful to use a router (or router firmware) that allows changes to Assured and Unreplied UDP timeouts for use with SIP services.