How To Make A Test Call With Freeswitch
Best practices for transcoding OPUS/G711 using freeswitch. Malwarebytes install with license key. Call Flow: SIP Client registered to a SIP Server dials a call to a PSTN number using the codec OPUS. SIP server proxy the call to Freeswitch with codec OPUS. Freeswitch terminates the call to SIP provider using G711u. Freeswitch does the transcoding between OPUS and G711u. Silence Supression is turned off on both legs. PTIME is 20 on both legs. However, for troubleshooting purposes, we requested the SIP clients to switch to G711u and all the test calls completed without any quality issues. I have installed Freeswitch 1.6 on RHEL6 on a Dell PowerEdge R710 with 16 cores and 96GB RAM. Call Flow: SIP Client registered to a SIP Server dials a call to a PSTN number using the codec OPUS.
Can someone share their experience of transcoding OPUS/G711 ans vice versa using Freeswitch? I am getting call quality issues even if there is a single call on the server. I am getting crackling noise and the end of the words.
Oct 31, 2017 I'd like to know how to make a test call to verify voice quality, etc. Using Skype for Business. I've tried the info at the below link, but this does not seem to work. When I search for 'Echo / Sound Test Service' using the Skype Directory tab, I get a ton of results, most of which look to be fake.
SIP Clients HAVE to use the OPUS, it is a customers requirements and there is nothing negotiable on this front. However, for troubleshooting purposes, we requested the SIP clients to switch to G711u and all the test calls completed without any quality issues.
I have installed Freeswitch 1.6 on RHEL6 on a Dell PowerEdge R710 with 16 cores and 96GB RAM.
Call Flow:SIP Client registered to a SIP Server dials a call to a PSTN number using the codec OPUS. SIP server proxy the call to Freeswitch with codec OPUS. Freeswitch terminates the call to SIP provider using G711u. Freeswitch does the transcoding between OPUS and G711u.
Silence Supression is turned off on both legs.PTIME is 20 on both legs.
Any suggestions would be much appreciated.
Tim1 Answer
use opus@8000h@20i with these settings in opus.conf.xml :
which direction do you have the audio issues ? it's important to know if there's a problem on the encoder or on the decoder.
Please file a jira here if you still have issues : https://freeswitch.org/jira/
Not the answer you're looking for? Browse other questions tagged freeswitchtranscoding or ask your own question.
I want to write a web app that connect to freeswitch and make outgoing call to some destination number (gateway for landline or internal sip devices) and play some sounds (may be do some logic in lua script).
After read freeswitch wiki ,I found originate command but it doesn't work for me (I just test for internal sip number - sofia/internal/username@ip ).if originate command can do this ,how to use it properly?if there is another way please tell me.
Thanks.
5 Answers
one way that I test and it work is run a lua script from freeswitch console or ESL:(ex 'luarun test.lua')
You can make the wav play when someone start a call, follow these steps.
- Place your wave into your freeswitch/conf folder.
Add the code bellow to your freeswitch/conf/autoload_configs
Run a HTTP server that receives a POST request and returns your dialplan(which tells freeswitch to play your wav).
- Make sure your freeswitch/conf/autoload_configs/xml_curl.conf.xml looks like this
Hope this helps.
How To Make A Test Call With Freeswitch Free
you can achieve By using a socket[ESL] application.
Originate command is used to make the call and bridge command is used to bridge the call.you can call originate command externally by using esl socket.
Examples:
refer this for esl written in node.jshttps://github.com/englercj/node-esl