Optimize a Freeswitch SIP Profile for Linphone

[danry25]

It might be best to create an optimized Freswitch SIPprofile for Linphone (eg: Lower the sample rate of Opus to 8000 from 48k, reduce bitrate, ensure forward error correction is maxed out and test different jitter buffer sizes).

[danry25]

I have created a SIP profile that has the following modified parameters:

Name	Value	Enabled	Reason
tls-only	true	true	Freeswitch was contacting Flexisip on port 5061 without using TLS
tls-bind-url	sips:mod_sofia@[IPv6 address Here]:5071;transport=tls	true	Trying to force SIPS scheme to be used rather than SIP, but did not work
inbound-codec-prefs	opus@48000h@20i,PCMU	true	Can be configured to force Opus, need to get Flexisip to handle transcoding to offload CPU use
multiple-registrations	false	true	Only permitting a single registration prevents spamming up the Linphone call log, Flexisip should handle call forking
auto-jitterbuffer-msec	60	true	Might enable Forward Error Correction (FEC) or Packet Loss Concealment depending on codec, sounds much better on Linphone & Grandstreams even when latency and jitter are minimal. 60ms setting has severe negative affect on Zoiper in call latency.

To do:

• Offload transcoding to Flexisip
• Make the Quit button unregister Linphone from Freeswitch
• Ask FusionPBX team about SIPS & SRTP (SRTP pending updating to latest FusionPBX)