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We were making some modifications to our interface to support multiple lines when we discovered a bug in hold/resume.
If you're on a call, and hold the call (button OR console command oSipSessionCall.hold(); ) it works fine. If you then resume the call (button OR console command oSipSessionCall.resume(); ) it also works fine. Then, if you try to re-hold the call, the hold button freezes and the entire sip stack begins acting strangely. The function calls seem correct, we enter the hold() code, but nothing happens, and we get a sipStack Event of "m_stream_audio_remote_added" instead of "m_local_hold_ok". We actually have to unregister and re-register to be able to start making calls again.
We reverted all changes we made and tested using the upstream library and experienced the same issue. Is anyone else experiencing this?
The text was updated successfully, but these errors were encountered:
Subsequent research gave more insight; we had been testing with a recording line, but upon testing with a real recipient we learned that the mic never wakes up on resume. The documented behavior is:
Start Call, both local audio and remote audio work.
Place call on hold, local and remote audio stop, the stack receives a hold_ok.
Resume the call, remote audio returns, local audio never starts sending, the stack receives a resume_ok.
Place call on hold again, remote audio remains, hold button never converts to "resume" button and remains in the disabled state.
We are researching the re-activation of the local audio stream as the potential culprit.
We were making some modifications to our interface to support multiple lines when we discovered a bug in hold/resume.
If you're on a call, and hold the call (button OR console command oSipSessionCall.hold(); ) it works fine. If you then resume the call (button OR console command oSipSessionCall.resume(); ) it also works fine. Then, if you try to re-hold the call, the hold button freezes and the entire sip stack begins acting strangely. The function calls seem correct, we enter the hold() code, but nothing happens, and we get a sipStack Event of "m_stream_audio_remote_added" instead of "m_local_hold_ok". We actually have to unregister and re-register to be able to start making calls again.
We reverted all changes we made and tested using the upstream library and experienced the same issue. Is anyone else experiencing this?
The text was updated successfully, but these errors were encountered: