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PortSIP PBX for Unified Communications

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  1. Today
  2. Hi Joe, I tried calling the PortSIP_playAudioFileToRemote method inside onInviteAnswered or onInviteConnected events, but both these events gets executed quite a while before the user picks up. Can you suggest any other event which I might be missing out or any other workaround....? Thanks, Fenil
  3. Yesterday
  4. Last week
  5. The flickering is not happening if only one audio codec is sent upon registration. It still happens every now and then, but it usually does not.
  6. The issue is solved if there is only a single audio codec sent to the PBX, that is what I've experienced just now, but I'll get back to you if I find anything else.
  7. This feature is not supported now. Our PortPBX V12.2 version , when create a extension, will Generate a QR code´╝Ü When the APP scan this QR code, can sign without enter the account. The QR code content is: { "display_name": "Ext 1001", "sip_domain": "test.com", "transports": [ { "protocol": "UDP", "port": "5060" } ], "pbx_public_ip": "", "pbx_private_ip": "", "email": "", "voicemail_number": "", "extension_number": "1001", "extension_password": "111111", "web_access_password": "123456" } If you use this format Generate a QR code, our softphone can scan it to login.
  8. Hello, is it possible to make a configuration file for the application,and pass it to a friend so that he simply goes to porsip and everything works for him?)
  9. can you capture the packet on your server, make a push call again, and send the capture file to me.
  10. This multiple INVITE is initiated by the server, I am not sure if the server will send multiple INVITE when there is only one codec, you can use one codec test again.
  11. When you start PortSIP_playAudioFileToRemote? Please start playAudioFileToRemote after onInviteAnswered or onInviteConnected
  12. Hi, do you using our PortSIP PBX or other PBX? In our PBX there will no " Timeout during media negotiation for call" error,
  13. I understand, thank you! So does this mean, that the PortSIP SDK handles all this stopping and starting of the audio session? What happens if I only advertise a single audio codec? Would that help getting rid of the flashing and the delay?
  14. Hello, Sure. The issue is the following: user A calls user B, when user B is not registered in PBX (closed the app for example) a VoIP push notification wakes the application of user B, which then registers with the PBX and receives the INVITE for some reason the media negotiation fails right after user B answers the invite Some additional information: user A registers with 3 audio codecs user B registers with the same 3 audio codecs after user B answers the call, on user A's side the onInviteUpdated is called with just one codec
  15. Hi Joe, Oohh God !! This was the mistake I've been doing since long. Thanks a lot Joe. Appreciated!! Now, I'm able to play the audio file on the remote device. Thanks for your consistent support. But, I would like to ask for your little help again. I'm able to play the audio file on remote device , but the issue is I can play the audio file even when the remote user hasn't picked up the call. And when the user picks up , half of the audio file has been played. I've implemented the event onInviteAnswered() but I can observe that the program call goes to this method even when the user hasn't picked up and the audio file starts playing. Could you please suggest some work around here ? Thanks & Best Regards, Fenil Shah
  16. would you please describe your questions more clear ? I don't understand your issue very clear.
  17. The messages are asynchronous, and we cannot control the order in which the messages arrive. You can add a successful registration event in the code - (void)onInviteIncoming­čś×long)sessionId callerDisplayName­čś×char *)callerDisplayName caller­čś×char *)caller calleeDisplayName­čś×char *)calleeDisplayName callee­čś×char *)callee audioCodecs­čś×char *)audioCodecs videoCodecs­čś×char *)videoCodecs existsAudio­čś×BOOL)existsAudio existsVideo­čś×BOOL)existsVideo sipMessage­čś×char *)sipMessage { NSLog(@"onInviteIncoming sessionId: %d", sessionId); [_portSIPHandle onRegisterSuccess:200 withStatusText­čś×char*)[@"" cStringUsingEncoding:kCFStringEncodingUTF8]]; }
  18. sendSdp to false, means INVITE message without SDP, 200 OK and ACK will Bring SDP. The reason for the microphone flashing is that the server has updated SDP, and the client needs to restart the audio channel. When the IOS system detects that the microphone is stopped, an icon will appear. If the server sends INVITE multiple times, there will be multiple flashes.
  19. Quick update: if I choose to set sendSdp to false when using the portSIPSDK.call method, then the call is connected even in the first described scenario. How should the call method be invoked?
  20. Quick update: if I choose to set sendSdp to false when using the portSIPSDK.call method, then the microphone flickering and delay is placed on the other side, to the callee's side. What could cause this? How can this be fixed?
  21. Hello, I am developing a VoIP application for iOS and I've run into the following problem: Caller is registered in the PBX Callee is not registered in the PBX Caller initiates outgoing call Callee receives a VoIP push notification, which starts the application in the background Callee performs the registration Callee receives the invitation even before the onRegisterSuccess callback Callee answers the call by invoking portSIPSDK.answerCall Callee gets 0 as the result of the previous call Caller instead of a 200 OK gets a 504 timeout Two notes on the above issue: If the same experiment is run while the callee is registered, then it works, the call gets connected The PBX logs state the following error: Timeout during media negotiation for call Thanks and regards, Lehel
  22. Actually, I was too fast to declare victory, there is still a bit of flickering and now a 2-3 second delay "only", but it doesn't seem as bad as with the other audio codec. I'll experiment with further options as well in this area. When I start a new call and it gets answered, from where can I retrieve the time of connection in order to sort of sync clocks?
  23. Usually when the call information changes, send INVITE again to update the SDP of the call Or for session-time check. Update SDP will recreate RTP channel, When the RTP channel is frequently recreated, the call audio may be stuck. i think the microphone icon flash Is this reason. I don't know why need this package
  24. Solved it! It seems, that the supported/negotiated audio codec was the problem, my application added AUDIOCODEC_OPUS as the first one and so that was what the negotiation with the callee and my app agreed upon. After I changed it to AUDIOCODEC_G722, the audio session issue disappeared instantly.
  25. 1. SDK can't check have an existing registration. 2. Even if there is an existing registration, we need to re-register if you open the program newly.
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