Closed icodelifee closed 2 years ago
Hi @icodelifee. Can You please test call via our test tool here:
You can log in with sip credentials and call your flutter app. It would be nice to confirm if in this case there is still no audio on both ends so we can isolate it to the flutter SDK.
@icodelifee I just tested one more time with the example app and it all seems to be working correctly.
Can you have a look at the reference application here: https://github.com/team-telnyx/telnyx-webrtc-flutter/tree/main/lib
It's a very simple implementation but should give you a good idea of how it works.
Do you have the correct permissions in the manifest?
Hi @icodelifee. Can You please test call via our test tool here:
You can log in with sip credentials and call your flutter app. It would be nice to confirm if in this case there is still no audio on both ends so we can isolate it to the flutter SDK.
I tried the webrtc app and was able to make calls using credentials but for my use case, we only require outbound calls
Yep i think i do have correct permissions
@icodelifee I just tested one more time with the example app and it all seems to be working correctly.
Can you have a look at the reference application here:
main
/libIt's a very simple implementation but should give you a good idea of how it works.
Do you have the correct permissions in the manifest?
I tried using the example app
Also facing this issue, even though i tried using the same organization number in webrtc.telnyx.com where it worked fine
Hi @icodelifee. Can You please test call via our test tool here: webrtc.telnyx.com You can log in with sip credentials and call your flutter app. It would be nice to confirm if in this case there is still no audio on both ends so we can isolate it to the flutter SDK.
I tried the webrtc app and was able to make calls using credentials but for my use case, we only require outbound calls
So in this case please log into that web app and then call the web app credentials from your flutter application
Also facing this issue, even though i tried using the same organization number in webrtc.telnyx.com where it worked fine
Please share the exact invite message sent on the socket
I tried using the example app
Are you on the Main branch? This error isn't happening for me currently. I will spend some time this weekend looking into this though.
There is a refactor that we have planned which will simplify this flow and might indirectly fix this issue
Also facing this issue, even though i tried using the same organization number in webrtc.telnyx.com where it worked fine
Please share the exact invite message sent on the socket https://katb.in/porozosiwor [Edited with full logs]
I tried using the example app
Are you on the Main branch? This error isn't happening for me currently. I will spend some time this weekend looking into this though.
There is a refactor that we have planned which will simplify this flow and might indirectly fix this issue
Yes im using the main branch
Also facing this issue, even though i tried using the same organization number in webrtc.telnyx.com where it worked fine
Please share the exact invite message sent on the socket https://katb.in/porozosiwor [Edited with full logs]
Have you enabled the +91 call code area within your portal?
I tried using the example app
Are you on the Main branch? This error isn't happening for me currently. I will spend some time this weekend looking into this though. There is a refactor that we have planned which will simplify this flow and might indirectly fix this issue
Yes im using the main branch
Strange, I will look in to this. You can't build a call without being logged in successfully which sets that sessionId. To me this sounds like somehow you aren't receiving a sessionId or logging in successfully before trying to create a call. Do you get a gateway registered log before attempting to make a call?
Also facing this issue, even though i tried using the same organization number in webrtc.telnyx.com where it worked fine
Please share the exact invite message sent on the socket katb.in/porozosiwor [Edited with full logs]
Have you enabled the +91 call code area within your portal?
I tried using the example app
Are you on the Main branch? This error isn't happening for me currently. I will spend some time this weekend looking into this though. There is a refactor that we have planned which will simplify this flow and might indirectly fix this issue
Yes im using the main branch
Strange, I will look in to this. You can't build a call without being logged in successfully which sets that sessionId. To me this sounds like somehow you aren't receiving a sessionId or logging in successfully before trying to create a call. Do you get a gateway registered log before attempting to make a call?
Yes i do receive the gateway log and yes +91 is enabled, i was able to make calls in my app
I'm in the Telnyx slack, if you would like you test with my credentials, maybe we could contact through there
Okay, I will look into this over the weekend and will do a relevant patch if required on Monday.
Hi @icodelifee. Can You please test call via our test tool here: webrtc.telnyx.com You can log in with sip credentials and call your flutter app. It would be nice to confirm if in this case there is still no audio on both ends so we can isolate it to the flutter SDK.
I tried the webrtc app and was able to make calls using credentials but for my use case, we only require outbound calls
So in this case please log into that web app and then call the web app credentials from your flutter application
i think we only have one telnyx number so i don't know if that will work or not
Okay, I will look into this over the weekend and will do a relevant patch if required on Monday.
Thank you
Hey, any update? @Oliver-Zimmerman
@icodelifee Fixed: https://github.com/team-telnyx/telnyx-webrtc-flutter/pull/14
Please test with telnyx_webrtc: ^0.0.4
@icodelifee Fixed: #14
Please test with telnyx_webrtc: ^0.0.4
Thank you for the update, I'm testing it right now and the mic and audio still doesnt work and also SocketMethod.BYE and RINGING is not being returned in the callback" logs: https://katb.in/ujoyituzayi
I had a chat with Telnyx representative the other day and i tried calling them through my app and i was able to hear and speak without any issue, i dont know why it doesnt work with other number
Hi @icodelifee thanks for the PR! Ringing doesn't necessarily need to be added because you should receive an INVITE message that functions the same, however I will review your PR and test it and merge if all is working.
In terms of BYE, what do you mean? You aren't getting a bye message when a call ends? It could be that the message isn't being interpreted by JSON as an actual BYE message. I can look into that
Hi @icodelifee thanks for the PR! Ringing doesn't necessarily need to be added because you should receive an INVITE message that functions the same, however I will review your PR and test it and merge if all is working.
In terms of BYE, what do you mean? You aren't getting a bye message when a call ends? It could be that the message isn't being interpreted by JSON as an actual BYE message. I can look into that
Hi, I did not receive an INVITE message when I tried calling a number, I needed RINGING for a specific use case.
I only receive bye when i endCall programmatically and not when I hang up the call on the receiving mobile. I want to know if the receiver hangup the call so I can update the UI .
@icodelifee can you confirm if this is calling a SIP Connection or an actual mobile number?
@icodelifee can you confirm if this is calling a SIP Connection or an actual mobile number?
actual number
Thanks, I'll look into this right now
Okay I can reproduce, I was testing before with SIP connections. I will look into a solution
Hey @Oliver-Zimmerman, i just tried the example app and I'm not able to hear anything
Hi @icodelifee, as mentioned before. I can reproduce this now and am working on it. Before I was testing sip connection to sip connection. The error is between sip connection and a real number.
I will update you when I can.
Hi, any updates? Thank you
Hi @icodelifee, we are still working on this issue. It's very strange and may be related to the backend. We are entirely sure why it is only affecting PSTN calls. I am hoping to have a more substantial update for you by the end of today
Hi @icodelifee, we have isolated the issue to something relating to the Flutter WebRTC library and how it is interacting with our backend. We are working to patch it at the moment. It involves Ice Candidates including a local host IP address. We are testing ignoring it as it comes in. Hoping to have it resolved today
@icodelifee we have a fix. I am opening a PR now. Once it is merged I will release a new version that contains your RINGING code as well
@icodelifee we have released 0.0.5 https://pub.dev/packages/telnyx_webrtc
Please test and close if it is working
Hey @Oliver-Zimmerman, I just tested and calls are working fine. Thank you. But is there a way to switch from speakerphone to earpiece speaker.
Hi @icodelifee, it is possible but would require a small bit of development. If you open a feature request I can add it to the backlog and notify you when it is complete.
Essentially we need to add a method that gets the current track and .enableSpearkerphone(true) or false. I will close this current issue for now. Please open a feature request instead.
Bug Category
SDK Version telnyx_webrtc: ^0.0.3
Describe the bug After the receiver accepts the call, both the caller nor the receiver cannot hear any audio from the device
Expected behavior The caller and receiver should be able to hear and speak
Device (please complete the following information):
Logs
INVITATION ANSWERED :: {jsonrpc: 2.0, id: 67246, method: telnyx_rtc.answer, params: {callID: Instance of 'Uuid', variables: {Event-Name: CHANNEL_DATA, Core-UUID: efaeab2c-2067-42d3-b973-37aa4330ac97, FreeSWITCH-Hostname: b2bua-rtc.tel-sy1-ibm-prod-133, FreeSWITCH-Switchname: b2bua-rtc.tel-sy1-ibm-prod-133, FreeSWITCH-IPv4: 10.33.0.80, FreeSWITCH-IPv6: ::1, Event-Date-Local: 2022-08-19 14:24:34, Event-Date-GMT: Fri, 19 Aug 2022 14:24:34 GMT, Event-Date-Timestamp: 1660919074493112, Event-Calling-File: switch_channel.c, Event-Calling-Function: switch_channel_get_variables_prefix, Event-Calling-Line-Number: 4576, Event-Sequence: 163053}}}