I have a browser-based WebRTC softphone setup using SipJS and FusionPBX. My FreeSwitch version is 1.10.2
If I call out from SipJS to an internal extension or an external pstn number, audio is working both ways.
If I call in to SipJS from a SIP softphone or and external pstn call then the SipJS client cannot hear anything (but the other end can)
If I make a SipJS to SipJS call, audio works both ways
SIP Softphone to Sip Softphone (grandstream app or Zoiper) works fine.
The server is on a public IP. I have tried incoming calls from behind a NAT and not behind a NAT.
SipJS client is using a google STUN server
There is no difference when using Firefox or Chrome.
I currently have the following ports open:
TCP: 22, 80, 443, 2855-2856, 3478-3479, 5066, 8081-8082
TCP/UDP: 5060, 5061, 5070, 5080, 5081, 7443
UDP: 16384-32768
I saw in the sipjs log some ports higher up, so I temporarily opened 45000-65500 just to see if that helped but it did not make a difference.
I have my ip address set in the variables and sip profiles and have followed every other guide/forum post I can find.
I have attached logs from both freeswitch cli and the sipjs console log.
Does anyone on here have any ideas on how to get the incoming audio stream to work or have some troubleshooting tips so I can trace the problem?
Part of me is leaning towards it being a codec issue, seems how it happens when I go from WebRTC to SIP, but when calling out I don't have the problem so that makes me think it is a firewall/routing issue.
If I call out from SipJS to an internal extension or an external pstn number, audio is working both ways.
If I call in to SipJS from a SIP softphone or and external pstn call then the SipJS client cannot hear anything (but the other end can)
If I make a SipJS to SipJS call, audio works both ways
SIP Softphone to Sip Softphone (grandstream app or Zoiper) works fine.
The server is on a public IP. I have tried incoming calls from behind a NAT and not behind a NAT.
SipJS client is using a google STUN server
There is no difference when using Firefox or Chrome.
I currently have the following ports open:
TCP: 22, 80, 443, 2855-2856, 3478-3479, 5066, 8081-8082
TCP/UDP: 5060, 5061, 5070, 5080, 5081, 7443
UDP: 16384-32768
I saw in the sipjs log some ports higher up, so I temporarily opened 45000-65500 just to see if that helped but it did not make a difference.
I have my ip address set in the variables and sip profiles and have followed every other guide/forum post I can find.
I have attached logs from both freeswitch cli and the sipjs console log.
Does anyone on here have any ideas on how to get the incoming audio stream to work or have some troubleshooting tips so I can trace the problem?
Part of me is leaning towards it being a codec issue, seems how it happens when I go from WebRTC to SIP, but when calling out I don't have the problem so that makes me think it is a firewall/routing issue.
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