Hi friends.
we have a comercial android app using SIP stack for voice. We want to migrate that technology to webRTC and WCS and in fact we have prepared a testbed with this characteristics:
- Android app using flashphoner android SDK
- WCS as webrtc to SIP gateway
- Asterisk
- External SBC (acting as SIP to TDM gateway)
When we make a call from the android app to public telephone number the flow sequence is as follow:
Android app with wifi 300mbs symmetric calls webrtc to WCS. In the same machine the WCS sends via SIP the call to local asterisk, an the asterisk sends the call to the SBC that deliver the call to a fix phone. The problem is that the stream from the fix phone to the app is having an upset latency :-(.
We discard any kind of problem in the network, asterisk or SBC side, since we don't have latency with the current SIP stack that folows exactly the same process. We are using G729.
How can we improve that latency that doesn't allow us to migrate the technology?
Thanks in advance.
Mario
we have a comercial android app using SIP stack for voice. We want to migrate that technology to webRTC and WCS and in fact we have prepared a testbed with this characteristics:
- Android app using flashphoner android SDK
- WCS as webrtc to SIP gateway
- Asterisk
- External SBC (acting as SIP to TDM gateway)
When we make a call from the android app to public telephone number the flow sequence is as follow:
Android app with wifi 300mbs symmetric calls webrtc to WCS. In the same machine the WCS sends via SIP the call to local asterisk, an the asterisk sends the call to the SBC that deliver the call to a fix phone. The problem is that the stream from the fix phone to the app is having an upset latency :-(.
We discard any kind of problem in the network, asterisk or SBC side, since we don't have latency with the current SIP stack that folows exactly the same process. We are using G729.
How can we improve that latency that doesn't allow us to migrate the technology?
Thanks in advance.
Mario