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tcz / rotary-gpt

I turned my old rotary phone into a ChatGPT-based home assistant and now you can too
MIT License
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Audio Get Code off #2

Closed maciasr closed 1 year ago

maciasr commented 1 year ago

hi there, hope all is well. I'm trying to setup but my audio is getting cut off. No a programmer, can you tell me where to add/edit the API key and the txt you suggested to add for GPT?

export OPENAI_API_KEY="xxxx" export AWS_ACCESS_KEY="yyyy" export AWS_SECRET_KEY="zzz"

Set it to the name of the city you leave. Used for weather.

export ROTARYGPT_PHYSICAL_LOCATION="Barcelona, Spain"

Getting the error below..

image

maciasr commented 1 year ago

Thank you for your help.

tcz commented 1 year ago

Hello,

It seems that the environmental variables are not set. Is that screenshot from a Windows machine? If so, I believe under Windows you need to use "set" and not "export", but I'm not an expert:

https://learn.microsoft.com/en-us/windows-server/administration/windows-commands/set_1

raulmaciascam commented 1 year ago

Thank you the environment variable did the trick. however I'm still facing issue with the granstream. I opened a ticket with them, below is a copy of the trace. I thank you very much.

rotarygpt3.zip

raulmaciascam commented 1 year ago

Grandstream claims that the issue since to be the RTP out port 5xxxx coming from the computer. I tried different versions of Windows. On W7 I'm getting partial response but breaks on W10 works a little better but breaks too. On W11 does not work at all. Just bought a raspberry pi will advise how it goes..

Below is grandstream respond..

Francisco @September 14, 2023 11:19

We suspect there is some misconfiguration at the rotaryGPT. It is sending RTP out of a port in 5xxxx, and is imposing the HT send to port 5004. The RTP port for sending and receiving ought to be the same one.

tcz commented 1 year ago

Can you please link to the full conversation with them? I'd like to see what you sent them.

dick-nds commented 1 year ago

I've the same issue with the Granstream HT801 (hardware version: v6.0A / software version: v1.0.49). Audio is (always) getting cut off during the greeting. Only "Hello" is audible, thenafter all following receiving audio (from the RTP queue) is ignored / trashed / etc.

Voice commands back to Rotary-GTP work fine; answers are created, sent back, but again not audible.

Then i've hooked up a Gigaset E630A GO, made a direct IP call to Rotary-GTP, and everything works right out of the box....

Something with the Granstream HT801 is incorrectly set / parameterized. Already tried several settings but without success so far.

tcz commented 1 year ago

I'm sorry you're having a trouble here. I suspect that the hardware version might be a culprit, yours is much newer than mine.

As I wrote in #1, I think it would be useful to run debugging on the HT801: https://www.loom.com/share/a480ec34037944649282994a1a42cf4a?sid=ce15ddba-bb28-4d7d-abea-d701c5e14cba

This might reveal some issues.

dick-nds commented 1 year ago

No problem. Thanks. I saw the debug option as well.

I will continue testing/debugging and/or hook up a Cisco ATA.

As for using Rotary-GPT with home automation.... Have you ever thought about creating an interface for Home Assistant (https://developers.home-assistant.io/docs/api/rest/)? That would be great!

tcz commented 1 year ago

I'm not familiar with Home Assistant, I only made connectors for stuff I own/use but contribs are welcome! Let me know if I can help with the debug! It's exciting to hear you're setting up RotaryGPT!

raulmaciascam commented 1 year ago

Grandstream Helpdesk.pdf

Attached is my interaction with Grandstream.

raulmaciascam commented 1 year ago

I've the same issue with the Granstream HT801 (hardware version: v6.0A / software version: v1.0.49). Audio is (always) getting cut off during the greeting. Only "Hello" is audible, thenafter all following receiving audio (from the RTP queue) is ignored / trashed / etc.

Voice commands back to Rotary-GTP work fine; answers are created, sent back, but again not audible.

Then i've hooked up a Gigaset E630A GO, made a direct IP call to Rotary-GTP, and everything works right out of the box....

Something with the Granstream HT801 is incorrectly set / parameterized. Already tried several settings but without success so far. Same exact device.

image

dick-nds commented 1 year ago

So, I've done some debugging with different logging levels... The most detailed debug logging level is quite cryptic, and no documentation for it is available anywhere. I saw some RTCP buffer issues. Which may disrupt the RTP setup process.

I disagree with Grandstream Helpdesk's response. Each side in a bidirectional RTP session assigns their source ports independently. Each side in a unicast session simply indicates to the other side where it wants to receive RTP packets.

Today I tested a Cisco ATA 191/192. This works instantly with factory default settings.

Also "off-hook auto dial" is possible with this ATA device. My dial plan setting is (where xxx are ip address decimals):

(P0 <:xxx.xxx.xxx.xxx:5060>)

For now I'm happy with (the more robust) Cisco ATA.

tusabez commented 1 year ago

So, I've done some debugging with different logging levels... The most detailed debug logging level is quite cryptic, and no documentation for it is available anywhere. I saw some RTCP buffer issues. Which may disrupt the RTP setup process.

I disagree with Grandstream Helpdesk's response. Each side in a bidirectional RTP session assigns their source ports independently. Each side in a unicast session simply indicates to the other side where it wants to receive RTP packets.

Today I tested a Cisco ATA 191/192. This works instantly with factory default settings.

Also "off-hook auto dial" is possible with this ATA device. My dial plan setting is (where xxx are ip address decimals):

(P0 <:xxx.xxx.xxx.xxx:5060>)

For now I'm happy with (the more robust) Cisco ATA.

Same issues with Grandstream so I tried using a Cisco ATA 192 and followed the same dial plan setting with my RPI's IP address but it doesn't seem to be calling out to my Raspberry PI. Anything else in the menu that I need to change for it to work?

dick-nds commented 1 year ago

Yes, i forgot to mention that you need to set "Make Call Without Reg:" to "Yes" in section "Proxy and Registration" settings for "Line 1" or "Line 2".

raulmaciascam commented 1 year ago

Hi, Thank you for your reply unfortunately I don't see that entry on FXS port session of the HT801 device.

[A screenshot of a computer Description automatically generated]

From: dick-nds @.> Sent: Thursday, September 21, 2023 4:22 AM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Manual @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

Yes, i forgot to mention that you need to set "Make Call Without Reg:" to "Yes" in section "Proxy and Registration" settings for "Line 1" or "Line 2".

- Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1729085407, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QB6EGAS5KU6E7FU4D3X3P2KVANCNFSM6AAAAAA4HWEL5A. You are receiving this because you are subscribed to this thread.Message ID: @.**@.>>

dick-nds commented 1 year ago

It's was mentioned for the Cisco ATA (in reply to @tusabez).

tcz commented 1 year ago

I disagree with Grandstream Helpdesk's response. Each side in a bidirectional RTP session assigns their source ports independently.

I agree with @dick-nds here, Grandstream's response is strange.

@raulmaciascam have you been able to debug HT801 as suggested in #1? I'd love to see that log.

raulmaciascam commented 1 year ago

I will try and post the results.

tusabez commented 1 year ago

@dick-nds That did the trick! Thanks! @tcz If I wanted to change the default voice to American English and pick a server in the Western United States, would I need to make changes just in aws.py? I'm still new to this type of project so trying to figure things out.

tcz commented 1 year ago

Yes, changing this line

self.target_host = "polly.eu-west-1.amazonaws.com"

Should change the region. I guess it could have been a parameter.

Regarding the voice, indeed, change this line:

default_voice = "Daniel"

It's important that you need to pick a voice that supports Neural Voice. Here's the full list: https://docs.aws.amazon.com/polly/latest/dg/voicelist.html

raulmaciascam commented 1 year ago

Hi

Can you share pictures of your ATA 191/192 settings. Got me one but cant get it to work.

From: Zoltan Toth-Czifra @.> Sent: Wednesday, September 27, 2023 1:54 PM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Mention @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

Yes, changing this line

self.target_host = "polly.eu-west-1.amazonaws.com"

Should change the region. I guess it could have been a parameter.

Regarding the voice, indeed, change this line:

default_voice = "Daniel"

It's important that you need to pick a voice that supports Neural Voice. Here's the full list: https://docs.aws.amazon.com/polly/latest/dg/voicelist.html

- Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1737839654, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QBAZTMQIWP57L664ADX4RR2NANCNFSM6AAAAAA4HWEL5A. You are receiving this because you were mentioned.Message ID: @.**@.>>

maciasr commented 1 year ago

Log from HT801 below..

<15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.485465 LIBGSDSP: CSS: RTPSTART: dec_chan=0, coderTypeDec = 46, status = 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.492825 LIBGSDSP: CSS: RTPSTART: enc_chan=1coderTypeEnc = 46, status = 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.493362 LIBGSDSP: CSS: Value of sid_update for G711 is 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.498513 LIBGSDSP: CSS: pLineData->dwCapabilities = 3146059, pLineData->dec_chan = 0, jib_instance = 180a <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.498866 LIBGSDSP: CSS: @@@@@ Value of pktsize 10 ms is 1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.499340 LIBGSDSP: CSS: @@ ### JB ADAPTATION TYPE is ENABLED <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.499553 LIBGSDSP: CSS: The MAX JIB value received from User space 1000 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.499748 LIBGSDSP: CSS: The ADAPTATION TYPE value received from User space 1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.504103 LIBGSDSP: CSS: The RESYNCRONISATION THRESHOLD received from User space 3 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.504476 LIBGSDSP: CSS: The TARGET PLAYOUT DELAY received from User space 20 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.507813 LIBGSDSP: CSS: The MONITORING INTERVAL value received from User space 2000 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.508364 LIBGSDSP: CSS: The ADAPTATION STEPSIZE RESET TIME value received from User space 60 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.508596 LIBGSDSP: CSS: The ADAPTATION SLOPE value received from User space 24 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.508799 LIBGSDSP: CSS: The POST ADAPTATION STEP SIZE MEDIUM value received from User space 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.509160 LIBGSDSP: CSS: The POST ADAPTATION STEP SIZE value received from User space 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.509396 LIBGSDSP: CSS: The MIN JIB value received from User space 120 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.509595 LIBGSDSP: CSS: The JIB has been initialized to this 23926b8 for line id 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.520238 LIBGSDSP: CSS: Function p_rtpapp_Start;Line 1848; STATE 2 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.520608 LIBGSDSP: CSS: Line variables: bBufSize: 160;bDecFrameSize: 80;bDecPackedFrameSize: 80;bFrameSize: 80;bPackedBufSize: 160, bPackedFrameSi <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.520748 LIBGSDSP: ze: 80 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.521155 LIBGSDSP: CSS: voice_request_start_chan returning 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.521414 LIBGSDSP: CSS: In API callback event = 281, inst = 1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.521627 LIBGSDSP: CSS: p_rtpapp_AUCCallback is getting called <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.521824 LIBGSDSP: CSS: p_rtpapp_AUCMssgHandler: Encoder chan 1 started for line 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.522176 LIBGSDSP: CSS: In API callback event = 281, inst = 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.528446 LIBGSDSP: CSS: p_rtpapp_AUCCallback is getting called <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.528730 LIBGSDSP: CSS: p_rtpapp_AUCMssgHandler: Decoder chan 0 started for line 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.529094 LIBGSDSP: CSS: p_rtpapp_SetSesionStatus: started decoder and encoder for line 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.529346 LIBGSDSP: CSS: ##### AUC Channel started sending COMA REPLY ##### <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.529556 LIBGSDSP: CSS: Function create_voice_message; Sending coma response 5 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.535437 LIBGSDSP: CSS: Function p_rtpapp_SetSesionStatus;Line 703; STATE CHANGE: to 3 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.538246 LIBGSDSP: CSS: voice_request_start_rtcp:756: Calling voice_start_rtcp <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.538535 LIBGSDSP: CSS: Function p_rtpapp_RtcpStart;Line 151; Calling p_rtcp_SessionInit <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.538751 LIBGSDSP: CSS: rtcp interval 5 and opts flag 19 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.539088 LIBGSDSP: CSS: ### p_rtcp_SessionInit 5 opts 19 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.544595 LIBGSDSP: CSS: The RTCP Feedback type is '0' fb_bw '0' fb_trr_interval '0' max_rtt 0 and sizeof RTCP_config 816 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.544880 LIBGSDSP: CSS: Session RB TYPE :: 255 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.545291 LIBGSDSP: CSS: AFTER COPY TO rtcpsession interval 5 opts 19 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.545511 LIBGSDSP: CSS: The RTCP XR - thining 3 & gmin 16 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.546221 LIBGSDSP: CSS: p_rtcp_xr_SessionInit function called ---- Number Of SIP SESSIONS '8' <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.546483 LIBGSDSP: CSS: p_rtcp_xr_SessionInit called with Gmin : 16 and Thining : 3 value <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.551348 LIBGSDSP: CSS: Modes of RTCP 19 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.551718 LIBGSDSP: CSS: ######## RTCP interval given from user space 5 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.552145 LIBGSDSP: CSS: p_rtcp_xr_SessionStart function called ---- <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.555197 LIBGSDSP: CSS: Function create_voice_message; Sending coma response 14 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.555492 LIBGSDSP: CSS: p_rtp_SeqInit 75 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364491.560415 LIBGSDSP: CSS: p_rtp_SeqInit 77 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.645840 SigCtrl::performRegistration on acct 0, Registered: 1, proxyUrl: 254.255.107.58.site.sil6.broadviewnet.net, preferPrimary:0, perfer Primary <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG OBP:0, obpIndex:0, failbackTimer: 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.651109 SigCtrl::performRegistration on acct 0, reset clear reachable flag to true <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.656127 SIPStack(0)::cb_snd_message, host=212232ABJJ.sil6.broadviewnet.net, original port=5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.656429 SIPStack(0)::cb_snd_message: t retry count =1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.656638 SIPStack(0)::snd_message, dst port:5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.656880 SIPStack(0)::snd_message: Copy registrarIP[] to registrarIPAddr: 216.214.56.86, port: 5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.659464 Line 452, DNS::lookup, Fetch DNS A record from preferred IP[0] for host 212232ABJJ.sil6.broadviewnet.net, IP: 216.214.56.86, port =5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.659785 SIPStack(0)::snd_message: UDP case. setNetworkConnected <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.660200 SIPStack(0)::snd_message: Present IP: 216.214.56.86, port: 5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.660427 SIPTransaction::setTargetIPandPort, host: 212232ABJJ.sil6.broadviewnet.net, IP: 216.214.56.86, port: 5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.660651 Line 452, DNS::lookup, Fetch DNS A record from preferred IP[0] for host 212232ABJJ.sil6.broadviewnet.net, IP: 216.214.56.86, port =5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.660836 SIPStack(0)::snd_message: Save registrarIP: 216.214.56.86, registrarPort: 5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.661730 SIPStack(0)::snd_message, update credential successful for server IP: 216.214.56.86 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364494.665323 SIPClientTransaction::sendRequest: Request 362 is sent <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.674130 SIPStack(0)::snd_message:(973)REGISTER sip:254.255.107.58.site.sil6.broadviewnet.net SIP/2.0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Via: SIP/2.0/UDP 254.255.107.58:5060;branch=z9hG4bK1904225867;rport Route: <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG From: "Tech Room" ;tag=121971311 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG To: Call-ID: 1658516088-5060-3@CFE.CFF.BAH.FI CSeq: 2345 REGISTER <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Contact: ;reg-id=1;+sip.instance="" <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Authorization: Digest username="3786", realm="732985AAAE", nonce="297381190", uri="sip:254.255.107.58.site.sil6.broadviewnet.net", response= <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG "0c9e743215ccd69a624519f5d9f989e7", algorithm=MD5, cnonce="08814139", qop=auth, nc=0000008e Max-Forwards: 70 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG User-Agent: Grandstream HT801 1.0.49.2 c074adca6f20 MAC: c0:74:ad:ca:6f:20 Supported: path Expires: 3600 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364494.686741 SIPStack(0)::run: Active call dialogs: 2 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364494.687223 SIPStack(0)::run: Active transactions: 1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.697119 SIPStack(0)::receiveMessage:(936)SIP/2.0 200 OK Via: SIP/2.0/UDP 254.255.107.58:5060;branch=z9hG4bK1904225867;rport <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Route: From: "Tech Room" ;tag=121971311 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG To: Call-ID: 1658516088-5060-3@CFE.CFF.BAH.FI CSeq: 2345 REGISTER <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Contact: ;reg-id=1;+sip.instance="";expires=30 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Authorization: Digest username="3786", realm="732985AAAE", nonce="297381190", uri="sip:254.255.107.58.site.sil6.broadviewnet.net", response= <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG "0c9e743215ccd69a624519f5d9f989e7", algorithm=MD5, cnonce="08814139", qop=auth, nc=0000008e Max-Forwards: 70 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG User-Agent: Grandstream HT801 1.0.49.2 c074adca6f20 MAC: c0:74:ad:ca:6f:20 Supported: path Expires: 30 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364494.711319 SIPStack(0)::cb_rcv2xx: Received 200 response for transaction 362(REGISTER), failoverInProgress=0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.714343 EventManager::run: Dispatching event 34 (SIG_REGISTERED) on port -1:-1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.715773 SigCtrl::setRegistered, acct:0, register status:1, voltage timer 0, delay effect time 0, uptime 5238, status 0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364494.723109 SIPTransaction::waitForResponse: Request 362 got status code 200 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.723767 SigCtrl::performRegistration on acct 0, regRetries:0, retryAfter:0, regRetryWait:1 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364494.725785 SIPStack(0)::run: Active call dialogs: 2 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364494.729467 SIPStack(0)::run: Active transactions: 1 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364494.735821 SigCtrl::processSigRegistered, Account 0 Registered, tried 0; Next reg in 15 seconds (5253) on 254.255.107.58.site.sil6.broadviewnet.net <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364494.737581 SigCtrl::processSigRegistered, SIP OPTIONS/NOTIFY Keep Alive starts for account 0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364494.739260 SigCtrl::performOptionsKeepAlive, SIP OPTIONS/NOTIFY Keep Alive SIP server at account 0 is alive <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.413226 LIBGSDSP: CSS: @@@## #$$$ Sending RTCP based on TS <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.417877 GSDSP::event_handler_rtp, port = 0, event = 3 data = -969971452, detected <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.418465 GSDSP::event_handler_rtp, port = 0, event = 3 data = -969971452, detected <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.420248 LIBGSDSP: CSS: p_rtcp_xr_report function called ---- <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.420600 LIBGSDSP: CSS: send Receiver ref. time block <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.420820 LIBGSDSP: CSS: Send Stat Summary RB <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.421206 LIBGSDSP: CSS: [p_rtcp_xr_StatWrite] we are having the loss packets of '200' <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.421432 LIBGSDSP: CSS: pkt_recvd '5' <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.421638 LIBGSDSP: CSS: We have exceed the size while writing the RTCP_XR_BT_LOSS_RLE <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.426661 LIBGSDSP: CSS: XR_LENGTH(wXRLen) '240' wRtcpXRSize '312' <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.427116 LIBGSDSP: CSS: ** Clear Session *** <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.429292 LIBGSDSP: CSS: ** p_rtcp_xr_ClearSession called ** <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.429588 LIBGSDSP: CSS: ** sizeof(pXRSession->stat.statsummary) :: 124 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.429807 LIBGSDSP: CSS: ** sizeof(pXRSession->stat.voip_mtr) :: 28 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.430180 LIBGSDSP: CSS: ** XR built successfully **Len =312 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.430465 LIBGSDSP: CSS: Sending RTCP pkt type '200' of length '52' to user space !!! <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.430674 LIBGSDSP: CSS: Sending RTCP pkt type '202' of length '20' to user space !!! <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.434349 LIBGSDSP: CSS: Sending RTCP pkt type '207' of length '240' to user space !!! <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.434601 LIBGSDSP: CSS: we are sending the rb RTCP_XR_VOIP_RB length '8' and size = '52' <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.434804 LIBGSDSP: CSS: rtp_event_send_notification called <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.435156 LIBGSDSP: CSS: we are not sending the rb RTCP_XR_DLRR_RB !!!!! <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.438160 LIBGSDSP: CSS: we are sending the rb RTCP_XR_STAT_RB length '9' and size '56' <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.438453 LIBGSDSP: CSS: rtp_event_send_notification called <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.438668 LIBGSDSP: CSS: we are not sending the rb RTCP_XR_RECREFT_RB!!!!! <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.438872 LIBGSDSP: CSS: we are not sending the rb RTCP_XR_LOSS_RLE_RB!!!!! <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.439259 LIBGSDSP: CSS: we are not sending the rb RTCP_XR_DUPLICATE_RLE_RB!!!!! <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.439481 LIBGSDSP: CSS: we are not sending the rb RTCP_XR_PKTRT_RB!!!!! <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364496.442733 LIBGSDSP: CSS: ### Sending RTCP XR RECV pkt to user space as sGRTCPQueryFlag[line] '9c0' <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.727845 SIPStack(0)::cb_nict_kill_transaction: Kill NICT transaction 362(REGISTER) <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.728405 SIPStack(0)::run: Active call dialogs: 2 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.737217 EventManager::run: Dispatching event 5 (PHONE_ON_HOOK) on port 0:0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.738238 ATACtrl::processPhoneOnHook on port 0:0, status = CALL_COMMUNICATION/CALL_IDLE isOffHook:1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.751861 Nuvoton::stopTone, Stop tone on port 0, direction 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.761007 Nuvoton::processEvent, Phone at port 0 is on-hook <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.773427 EventManager::run: Dispatching event 28 (CALL_COMPLETED) on port 0:0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.775511 SigCtrl::processCallCompleted on port 0:0, status = CALL_HANGUP/CALL_IDLE <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.780158 SigCtrl::processCallCompleted on port 0:0, Sending BYE to dialog 4, state=2 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.783078 SIPStack(0)::cb_snd_message, host=192.168.5.118, original port=5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.783402 SIPStack(0)::cb_snd_message: t retry count =1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.785665 SIPStack(0)::snd_message: Copy registrarIP[] to registrarIPAddr: 216.214.56.86, port: 5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.786139 SIPStack(0)::snd_message: Present IP: 192.168.5.118, port: 5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.786386 SIPTransaction::setTargetIPandPort, host: 192.168.5.118, IP: 192.168.5.118, port: 5060 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.790599 SIPClientTransaction::sendRequest: Request 363 is sent <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.795557 SIPStack(0)::snd_message:(537)BYE sip:192.168.5.118:5060 SIP/2.0 Via: SIP/2.0/UDP 254.255.107.58:5060;branch=z9hG4bK553948257;rport <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG From: "Tech Room" ;tag=1869409537 To: Call-ID: 1509639674-5060-7@CFE.CFF.BAH.FI <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG CSeq: 43 BYE Contact: Max-Forwards: 70 Supported: replaces, path, timer, eventlist <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG User-Agent: Grandstream HT801 1.0.49.2 c074adca6f20 MAC: c0:74:ad:ca:6f:20 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.805327 SigCtrl::processCallCompleted on port 0:0, Sending BYE to dialog 6, state=2 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.806832 SIPStack(0)::run: Active call dialogs: 2 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.810390 SIPClientTransaction::sendRequest: Request 364 is sent <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.816191 SIPStack(0)::receiveMessage:(289)SIP/2.0 200 OK Via: SIP/2.0/UDP 254.255.107.58:5060;branch=z9hG4bK553948257;rport <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG To: From: "Tech Room" ;tag=1869409537 Contact: <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Call-ID: 1509639674-5060-7@CFE.CFF.BAH.FI CSeq: 43 BYE Content-Length: 0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.834730 ATACtrl::processCallCompleted on port 0:0, status = CALL_IDLE/CALL_IDLE canConf:1 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.839777 SIPStack(0)::cb_rcv2xx: Received 200 response for transaction 363(BYE), failoverInProgress=0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.844227 SIPStack(0)::cb_snd_message, host=192.168.5.118, original port=5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.846323 SIPStack(0)::cb_snd_message: t retry count =1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.848142 SIPStack(0)::snd_message: Copy registrarIP[] to registrarIPAddr: 216.214.56.86, port: 5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.849376 Call(8)::processEvent, CALL_COMPLETED, port 0:0, evtPort 0:0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.849806 ### Line 2929, Call(8)::processMedia on port 0:0, stopping RTP, setting noEarlyRtpStop to false <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.850403 ATACtrl::stopRTP on 0:0, status = CALL_IDLE, force: 0, specialFeature = 100 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.851097 Call(8)::processEvent, CALL_COMPLETED, port 0:0, evtPort 0:0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.863350 GSDSP::stop RTP on port 0:0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.866674 LIBGSDSP: CSS: voice_request_stop_chan <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.868193 LIBGSDSP: CSS: voice_stop_chan <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.868509 LIBGSDSP: CSS: Sending RTCP BYE Packet <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.868731 LIBGSDSP: CSS: #### In function ByeLen reasonlen 3 pad 0 and rtcpLen 12 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.869076 LIBGSDSP: CSS: p_rtcp_ByeLen: 12 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.869325 LIBGSDSP: CSS: Sending RTCP pkt type '203' of length '12' to user space !!! <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.869526 LIBGSDSP: CSS: Function p_rtpapp_Stop;Line 2410; STATE CHANGE: to 4 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.876254 LIBGSDSP: CSS: pMember->stats.dwJibDuplicated '0' pMember->stats.dwJibDiscarded '0' pMember->stats.dwJibInvalid '0' pMember->stats.dwJib <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.876487 LIBGSDSP: Received '86', pMember->sta <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.880326 SIPStack(0)::snd_message: Present IP: 192.168.5.118, port: 5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.880671 SIPTransaction::setTargetIPandPort, host: 192.168.5.118, IP: 192.168.5.118, port: 5060 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.882150 EventPvalueChanged( :status_0_0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.885086 EventPvalueChanged( :4901 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.887183 EventPvalueChanged( :status_0_0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.890309 LIBGSDSP: dua_disconnect_voip:4643 ( 0, 0, 0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.892692 LIBGSDSP: dua_disconnect_voip:4695 ( 0, 0, 0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.894682 LIBGSDSP: dua_free_voip( 0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.907249 LIBGSDSP: dua_free_voip done <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.915894 LIBGSDSP: CSS: return value of p_rtpapp_Stopis 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.924275 EventManager::run: Dispatching event 64 (PVALUE_CHANGED) on port -1:-1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.928364 SigCtrl::processPvalueChange ( status_0_0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.930716 ATACtrl::processPvalueChange ( status_0_0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.934245 LIBGSDSP: dua_disconnect_fxs ( 0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.938595 LIBGSDSP: dua_disconnect_fxs done <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.944289 GSDSP::RTP stopped on port 0:0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.946463 RTP stop on port 0:0 local rtp port:5004 sdp:0x1d2948 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.948300 RTP::closeSocket(), Closing socket: 11, local RTP port: 5004 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.951213 RTP::closeSocket(), Closing socket: 21, local RTCP port: 5005 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.954453 Call(8)::processMedia, Call stopped on port 0:0, inTransfer = 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.955807 SIPStack(0)::snd_message:(538)BYE sip:192.168.5.118:5060 SIP/2.0 Via: SIP/2.0/UDP 254.255.107.58:5060;branch=z9hG4bK1622170837;rport <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG From: "Tech Room" ;tag=1817868183 To: Call-ID: 955313853-5060-11@CFE.CFF.BAH.FI <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG CSeq: 81 BYE Contact: Max-Forwards: 70 Supported: replaces, path, timer, eventlist <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG User-Agent: Grandstream HT801 1.0.49.2 c074adca6f20 MAC: c0:74:ad:ca:6f:20 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.969386 SIPStack(0)::run: Active call dialogs: 2 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.971199 EventManager::run: Dispatching event 64 (PVALUE_CHANGED) on port -1:-1 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364499.974196 ATACtrl Loop current disconnect skipped on port 0 because of onhook <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.978520 Call(8)::processMedia call_time_total total: 9 fxs:0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.981622 SigCtrl::processPvalueChange ( 4901 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.988829 ATACtrl::processPvalueChange ( 4901 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.991133 EventManager::run: Dispatching event 64 (PVALUE_CHANGED) on port -1:-1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364499.998484 SigCtrl::processPvalueChange ( status_0_0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.004555 ATACtrl::processPvalueChange ( status_0_0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.008715 EventPvalueChanged( :call_time_total_fxs0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.011453 EventManager::run: Dispatching event 64 (PVALUE_CHANGED) on port -1:-1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.014261 SigCtrl::processPvalueChange ( call_time_total_fxs0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.015549 ATACtrl::processPvalueChange ( call_time_total_fxs0 ) <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.017666 CallRecord::writeCDRFile, cdrBuf =3786,192.168.5.118:5060,N/A,2023-10-03 15:21:30,2023-10-03 15:21:31,2023-10-03 15:21:39,9,Normal,Outgoing <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG , len =106 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.022616 LIBGSDSP: CSS: In API callback event = 283, inst = 1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.026502 LIBGSDSP: CSS: p_rtpapp_AUCCallback is getting called <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.029894 LIBGSDSP: CSS: p_da_AUCChannelRelease status 0, ch 1 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.031391 LIBGSDSP: CSS: In API callback event = 283, inst = 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.038686 LIBGSDSP: CSS: p_rtpapp_AUCCallback is getting called <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.040451 LIBGSDSP: CSS: p_da_AUCChannelRelease status 0, ch 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.041792 LIBGSDSP: CSS: Function p_rtp_session_jb_stop;Line 2540; STATE CHANGE: to 5 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.044251 LIBGSDSP: CSS: p_rtp_SessionDestroy: 022e0660 - 022e3dd8 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.045826 LIBGSDSP: CSS: p_rtcp_xr_SessionStop function called ---- <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.048248 LIBGSDSP: CSS: ** Stat Summary **** <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.052639 CallRecord::writeCDRFile, writing /data/cdrc074adca6f20.csv OK. cdrBuf len =106 <14> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.INFO 1696364500.055242 Deleting Call object 8 port 0:0, callCount=0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.056608 EventManager::unregisterEventListener: listener Call <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.060317 RTP::~RTP, destruct 0x1ce168 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.062648 LIBGSDSP: CSS: ** Begin Seq :: 140 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.075074 LIBGSDSP: CSS: ** End Seq :: 1ec <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.079794 LIBGSDSP: CSS: ** Lost Packet :: ad <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.086239 LIBGSDSP: CSS: ** Du Packet :: 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.092816 LIBGSDSP: CSS: ** Max. Jitter value :: 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.094861 LIBGSDSP: CSS: ** Min. Jitter value :: 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.099527 LIBGSDSP: CSS: ** Mean. Jitter value :: 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.102775 LIBGSDSP: CSS: ** Max. TTL :: 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.104787 LIBGSDSP: CSS: ** Min. TTL :: 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.108599 LIBGSDSP: CSS: ** Mean. TTL :: 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.110156 LIBGSDSP: CSS: ** p_rtcp_xr_ClearSession called ** <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.113355 LIBGSDSP: CSS: ** sizeof(pXRSession->stat.statsummary) :: 124 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.118143 LIBGSDSP: CSS: ** sizeof(pXRSession->stat.voip_mtr) :: 28 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.119540 LIBGSDSP: CSS: total fb requested '0' and total removal of fb requested '0' <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.120783 LIBGSDSP: CSS: p_rtcp_SessionFree: 022e0760 - 022e12e0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.124187 LIBGSDSP: CSS: p_rtp_SessionDestroy: 022e0660 - 022e3dd8 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.128693 LIBGSDSP: CSS: ##### AUC Channel STOPPED sending COMA REPLY ##### <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.130421 LIBGSDSP: CSS: Function create_voice_message; Sending coma response 7 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.131654 LIBGSDSP: CSS: create_voice_message ret 0 <15> HT801 [c0:74:ad:ca:6f:20] [1.0.49.2] GS_ATA: USER.DEBUG 1696364500.139705 LIBGSDSP: CSS: Function create_voice_message; Sending coma response 9
maciasr commented 1 year ago

Got the Cisco device recommended no voice in either direction. Grandstream device I got voice outgoing but not incoming. I can see the replies from ChatGPT on the console but cant hear..

Mikklo commented 1 year ago

I am having the same issue with my HT801 (Hardware version: V1.1C). The sound is cut off during the greeting. No sound on polly playback. Mic works the whole time and I can see the response in the console but no audio playback after the greeting is cut short.

I can also add that it doesn't always cut out at the same time. I have a custom greeting.pcm that is a bit longer. A few times I've got the complete message, but most of the time it cuts mid-message.

I have tried with a software phone, MicroSIP, then I can hear the full greeting every time, but I still have not managed to get it to detect silence when calling from that program, so I have not been able to verify that polly sounds works.

tcz commented 1 year ago

@maciasr is that log from a RotaryGPT call? What's "Tech Room"

@Mikklo I'd maybe try to change the Jitter Buffer Length setting to High under the FXS PORT tab on the HT801 config but this really shouldn't be an issue.

raulmaciascam commented 1 year ago

hi,

yes I used the video that you provided to get the logs. At this point I think is a network problem seems the Grandstream device works one way and the cisco device not at all. I bought Raspberry pi, but still the same issue. I also bought the cisco device to tried but it did not work either. I still see the same message "polly sending to rtp" but no audio is heard.

Regards, Raul

From: Zoltan Toth-Czifra @.> Sent: Wednesday, October 4, 2023 5:22 PM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Mention @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

@maciasrhttps://github.com/maciasr is that log from a RotaryGPT call? What's "Tech Room"

@Mikklohttps://github.com/Mikklo I'd maybe try to change the Jitter Buffer Length setting to High under the FXS PORT tab on the HT801 config but this really shouldn't be an issue.

— Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1747660768, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QAWNFK63XMOZJKMTHLX5XHQRAVCNFSM6AAAAAA4HWEL5CVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMYTONBXGY3DANZWHA. You are receiving this because you were mentioned.Message ID: @.***>

Mikklo commented 1 year ago

@tcz Yeah, already tried that. I have a bit newer software version so not all settings match your screenshots, but I have matched them as far as possible and then played around with the extra settings I have that aren't in your screenshots but no success. I was really hoping to find a solution, so I don't have to go and buy another APA.

tcz commented 1 year ago

The reason it's so hard for me to debug this is that this seems to be an issue with the later hardware versions and it works perfectly on my earlier HT801. I looked at buying a new one but they only sell HT802 here for some reason.

I can mail you my HT801 and see if it works with that to discard other stuff like network.

raulmaciascam commented 1 year ago

Wow, that's so kind of you. I will mail it back to you or mail you a new one, just let me know your address. My address is below.

Raul Macias 45 Broadway, 25th Floor New York, NY 10006

From: Zoltan Toth-Czifra @.> Sent: Thursday, October 5, 2023 7:53 AM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Mention @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

The reason it's so hard for me to debug this is that this seems to be an issue with the later hardware versions and it works perfectly on my earlier HT801. I looked at buying a new one but they only sell HT802 here for some reason.

I can mail you my HT801 and see if it works with that to discard other stuff like network.

- Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1748733138, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QAMCPXIFMEFUEQKCO3X52NSFAVCNFSM6AAAAAA4HWEL5CVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMYTONBYG4ZTGMJTHA. You are receiving this because you were mentioned.Message ID: @.**@.>>

raulmaciascam commented 1 year ago

Where do you configure auto dial on Cisco ATA 191/192?

From: RaspiPico Noob @.> Sent: Wednesday, September 20, 2023 7:03 PM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Manual @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

So, I've done some debugging with different logging levels... The most detailed debug logging level is quite cryptic, and no documentation for it is available anywhere. I saw some RTCP buffer issues. Which may disrupt the RTP setup process.

I disagree with Grandstream Helpdesk's response. Each side in a bidirectional RTP session assigns their source ports independently. Each side in a unicast session simply indicates to the other side where it wants to receive RTP packets.

Today I tested a Cisco ATA 191/192. This works instantly with factory default settings.

Also "off-hook auto dial" is possible with this ATA device. My dial plan setting is (where xxx are ip address decimals):

(P0 <:xxx.xxx.xxx.xxx:5060>)

For now I'm happy with (the more robust) Cisco ATA.

Same issues with Grandstream so I tried using a Cisco ATA 192 and followed the same dial plan setting with my IP address but it doesn't seem to be calling out to my Raspberry PI. Anything else in the menu that I need to change for it to work?

- Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1728532906, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QFKYUAULPLKNHCHUPDX3NY37ANCNFSM6AAAAAA4HWEL5A. You are receiving this because you are subscribed to this thread.Message ID: @.**@.>>

raulmaciascam commented 1 year ago

Hi, any setting that we can modify to fixed that as it works on old HT801 devices.

Thank you,

From: Grandstream Help Desk @.> Sent: Thursday, October 5, 2023 4:58 PM To: Raul Macias @.> Subject: Our Engineers have added a comment to Ticket 20230908155004

There is a reply for Ticket 20230908155004

Dear Raul Macias,

Our engineers have replied to ticket 20230908155004.

Ticket Title: Audio Gets cuts off

Reply:

The port that rotary app proposes is 5004, and the audio is coming out of that app on port 55995. This the HT accepts in the beginning but then it locks on listening on audio coming from 5004 only.

Please check the ticket HEREhttps://helpdesk.grandstream.com/users/show_ticket/333116.

*** Please do not respond to this email regarding the support ticket, please respond directly through your HelpDesk account.

Thanks and have a great day!

Mikklo commented 1 year ago

@tcz I'm no expert on python and don't really know anything about RTP traffic. But I have a hard time seeing how it would end with audio on port 55995 as per grandstreams response to @raulmaciascam Could it somehow be that newer versions of HT801 somehow doesn't read the body of the 200 response properly and doesn't get the m=audio 5004 RTP/AVP 0 part where port 5004 is specified?

I will have to try and get debug logging working properly during the weekend. My debug log looks more like what @maciasr posted and not like the logs you get in your video. It does not show the 200 response from the RotaryGPT application at all.

raulmaciascam commented 1 year ago

Please see attached Wireshark capture. Grandstream HT801 device 192.168.10.169 SIP server=192.168.10.176 RTP traffic keeps selecting random ports to communicate from the SIP server. Any way to force the server to only use port 5004 to communicate to Ht801?

From: Mikklo @.**@.>> Sent: Friday, October 6, 2023 9:40 AM To: tcz/rotary-gpt @.**@.>> Cc: Raul Macias @.**@.>>; Mention @.**@.>> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

@tczhttps://github.com/tcz I'm no expert on python and don't really know anything about RTP traffic. But I have a hard time seeing how it would end with audio on port 55995 as per grandstreams response to @raulmaciascamhttps://github.com/raulmaciascam Could it somehow be that newer versions of HT801 somehow doesn't read the body of the 200 response properly and doesn't get the m=audio 5004 RTP/AVP 0 part where port 5004 is specified?

I will have to try and get debug logging working properly during the weekend. My debug log looks more like what @maciasrhttps://github.com/maciasr posted and not like the logs you get in your video. It does not show the 200 response from the RotaryGPT application at all.

- Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1750700890, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QFCXUGBLJWDUKOGBP3X6AC33AVCNFSM6AAAAAA4HWEL5CVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMYTONJQG4YDAOBZGA. You are receiving this because you were mentioned.Message ID: @.**@.>>

tcz commented 1 year ago

Grandstream's answer makes very little sense to me.

See the comment from @dick-nds here: https://github.com/tcz/rotary-gpt/issues/2#issuecomment-1726595755

RTP is a full-duplex protocol, meaning there's a separate socket for receiving as well as sending audio. Each UDP socket is defined by a two IP-address-port pairs. One of the receiver and the other one of the sender. The sender may assign an ephemeral port (meaning randomly chosen, usually from the higher port numbers) to send the data out of. The receiver usually assigns a fix port so that the sender knows where to send to. The sender's ephemeral port is typically transparently assigned by the operating system, since usually there's no reason to choose this manually at all.

An example socket could be: [192.168.1.2][5555] <--- [192.168.1.3][56789]

In case of RotaryGPT, we have two sockets (again, RTP is full-duplex):

During the SIP session both sides exchange port numbers using Session Description Protocol (SDP). The port numbers they exchange are the ones they expect the other party to connect to. It has nothing to do with the ephemeral ports used by the socket.

HT801 sends an INVITE SIP request to the RaspberryPi and in it it includes the protocols and ports it accepts the connection to (using the format defined by SDP). RotaryGPT parses this and extracts HT801's port to connect to, in this format:

r"audio (\d+) RTP"

This is the port RTPSender will connect to.

It also sends back its own port where it expect HT801 to connect. It case of RotaryGPT it's fix 5004. This is the port RTPReceiver will listen on.

Nowhere in this process the ephemeral ports play a part. This is why I'm very confused about Grandstream's response.

raulmaciascam commented 1 year ago

FYI- Grandstream just attached this two files to the case no explanation.

From: Zoltan Toth-Czifra @.> Sent: Friday, October 6, 2023 10:27 AM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Mention @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

Grandstream's answer makes very little sense to me.

See the comment from @dick-ndshttps://github.com/dick-nds here:

2 (comment)https://github.com/tcz/rotary-gpt/issues/2#issuecomment-1726595755

RTP is a full-duplex protocol, meaning there's a separate socket for receiving as well as sending audio. Each UDP socket is defined by a two IP-address-port pairs. One of the receiver and the other one of the sender. The sender may assign an ephemeral port (meaning randomly chosen, usually from the higher port numbers) to send the data out of. The receiver usually assigns a fix port so that the sender knows where to send to. The sender's ephemeral port is typically transparently assigned by the operating system, since usually there's no reason to choose this manually at all.

An example socket could be: [192.168.1.2][5555] <--- [192.168.1.3][56789]

In case of RotaryGPT, we have two sockets (again, RTP is full-duplex):

During the SIP session both sides exchange port numbers using Session Description Protocol (SDP). The port numbers they exchange are the ones they expect the other party to connect to. It has nothing to do with the ephemeral ports used by the socket.

HT801 sends an INVITE SIP request to the RaspberryPi and in it it includes the protocols and ports it accepts the connection to (using the format defined by SDP). RotaryGPT parses this and extracts HT801's port to connect to, in this format:

r"audio (\d+) RTP"

This is the port RTPSender will connect to.

It also sends back its own port where it expect HT801 to connect. It case of RotaryGPT it's fix 5004. This is the port RTPReceiver will listen on.

Nowhere in this process the ephemeral ports play a part. This is why I'm very confused about Grandstream's response.

— Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1750773588, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QGZNKXRXQ2GOGV6BATX6AIJ5AVCNFSM6AAAAAA4HWEL5CVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMYTONJQG43TGNJYHA. You are receiving this because you were mentioned.Message ID: @.**@.>>

raulmaciascam commented 1 year ago

From Grandstream..

There is a reply for Ticket 20230908155004

Dear Raul Macias,

Our engineers have replied to ticket 20230908155004.

Ticket Title: Audio Gets cuts off

Reply:

The screenshot you provided highlights the audio stream from HT to SIP server. This obeys the SDP negotiation.

The streams that do not obey SDP negotiation are the ones coming from 192.168.10.176 (SIP server, Raspberry...) . Please see attached.

Please check the ticket HEREhttps://helpdesk.grandstream.com/users/show_ticket/333116.

*** Please do not respond to this email regarding the support ticket, please respond directly through your HelpDesk account.

Thanks and have a great day!

From: Zoltan Toth-Czifra @.> Sent: Friday, October 6, 2023 10:27 AM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Mention @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

Grandstream's answer makes very little sense to me.

See the comment from @dick-ndshttps://github.com/dick-nds here:

2 (comment)https://github.com/tcz/rotary-gpt/issues/2#issuecomment-1726595755

RTP is a full-duplex protocol, meaning there's a separate socket for receiving as well as sending audio. Each UDP socket is defined by a two IP-address-port pairs. One of the receiver and the other one of the sender. The sender may assign an ephemeral port (meaning randomly chosen, usually from the higher port numbers) to send the data out of. The receiver usually assigns a fix port so that the sender knows where to send to. The sender's ephemeral port is typically transparently assigned by the operating system, since usually there's no reason to choose this manually at all.

An example socket could be: [192.168.1.2][5555] <--- [192.168.1.3][56789]

In case of RotaryGPT, we have two sockets (again, RTP is full-duplex):

During the SIP session both sides exchange port numbers using Session Description Protocol (SDP). The port numbers they exchange are the ones they expect the other party to connect to. It has nothing to do with the ephemeral ports used by the socket.

HT801 sends an INVITE SIP request to the RaspberryPi and in it it includes the protocols and ports it accepts the connection to (using the format defined by SDP). RotaryGPT parses this and extracts HT801's port to connect to, in this format:

r"audio (\d+) RTP"

This is the port RTPSender will connect to.

It also sends back its own port where it expect HT801 to connect. It case of RotaryGPT it's fix 5004. This is the port RTPReceiver will listen on.

Nowhere in this process the ephemeral ports play a part. This is why I'm very confused about Grandstream's response.

— Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1750773588, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QGZNKXRXQ2GOGV6BATX6AIJ5AVCNFSM6AAAAAA4HWEL5CVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMYTONJQG43TGNJYHA. You are receiving this because you were mentioned.Message ID: @.**@.>>

tcz commented 1 year ago

I don't think Github picks up your email attachments.

raulmaciascam commented 1 year ago

ip.addr==192.168.10.169.zip

raulmaciascam commented 1 year ago

You are correct, I just attached manually on the site. It's packet capture.

HT801 ip = 192.168.10.169 SIP server ip = 192.168.10.176

From: Zoltan Toth-Czifra @.> Sent: Friday, October 6, 2023 10:45 AM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Mention @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

I don't think Github picks up your email attachments.

- Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1750803523, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QAFC4TGI3H4K5F3GLDX6AKNXAVCNFSM6AAAAAA4HWEL5CVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMYTONJQHAYDGNJSGM. You are receiving this because you were mentioned.Message ID: @.**@.>>

raulmaciascam commented 1 year ago

1st_call_on_10_6_pcap_RTP_from_36383 2nd_call_on_10_6_pcap_RTP_from_49849

raulmaciascam commented 1 year ago

From Grandstream..

++++++++++++++++++++++++++++++++++

The screenshot you provided highlights the audio stream from HT to SIP server. This obeys the SDP negotiation.

The streams that do not obey SDP negotiation are the ones coming from 192.168.10.176 (SIP server, Raspberry...) . Please see attached.

tcz commented 1 year ago

Honestly I think they somehow get confused here and want to use the same socket for both directions. It's certainly possible with UDP but it's unusual and that's not how RTP is usually implemented.

My hunch is that the source of their confusion is perhaps the fact that we suggest using the same incoming port at RotaryGPT as they use on HT801. Perhaps in the newer hardware there's some "heuristics" that if the SIP server suggests the same port as the client then it has to be the same socket for audio in and out. This would be super odd and I still don't really get how and why they "cut" the communication when this happens.

My suggestion to verify this theory is to change the port of RotaryGPT:

port_change.patch

Do these two changes and please test again. Wireshark captures are useful if it still doesn't work.

raulmaciascam commented 1 year ago

image

raulmaciascam commented 1 year ago

hi,

My apologies, don't know much about programming, getting the error above. I'm my suppose to run the file using "python port_change.path" please see the error Im getting. I put the files on the same directory.

image

tcz commented 1 year ago

You need to apply the patch first:

git apply port_change.path

Then run RotaryGPT as usual.

raulmaciascam commented 1 year ago

Thank you, just tried with the update and encounter the same issue. Attached are the new wireshark captures.

raulmaciascam commented 1 year ago

wireshark-ip.addr==192.168.10.169.zip

tcz commented 1 year ago

Seems like this did nothing. Would you mind sending my response to Grandstream to see what they say to this? https://github.com/tcz/rotary-gpt/issues/2#issuecomment-1750773588

Maybe they will suggest using the same socket, I just don't really get why. Especially because it works well on the old hardware.

raulmaciascam commented 1 year ago

Zoltan-

I did, they usually take 24 hours to answer. I appreciate the effort. Thank you very much.

Raul

From: Zoltan Toth-Czifra @.> Sent: Friday, October 6, 2023 12:17 PM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Mention @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

Seems like this did nothing. Would you mind sending my response to Grandstream to see what they say to this?

2 (comment)https://github.com/tcz/rotary-gpt/issues/2#issuecomment-1750773588

Maybe they will suggest using the same socket, I just don't really get why. Especially because it works well on the old hardware.

- Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1751030988, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QFLKY6IFE4A7VAHCODX6AVIRAVCNFSM6AAAAAA4HWEL5CVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMYTONJRGAZTAOJYHA. You are receiving this because you were mentioned.Message ID: @.**@.>>

raulmaciascam commented 1 year ago

Zoltan-

Reply from grandstream to your questions.

Regards,

Raúl

On Oct 6, 2023, at 4:52 PM, Grandstream Help Desk @.***> wrote:

 There is a reply for Ticket 20230908155004

Dear Raul Macias,

Our engineers have replied to ticket 20230908155004.

Ticket Title: Audio Gets cuts off

Reply:

  1. The ports for the RTP sessions are the ones negotiated at the SDP level
  2. That is, since 5004 is the proposed by Rotary side, the rotary needs to send and receive audio on that port
  3. Please provide the section of a SIP RFC that requires otherwise

Please check the ticket HEREhttps://helpdesk.grandstream.com/users/show_ticket/333116.

*** Please do not respond to this email regarding the support ticket, please respond directly through your HelpDesk account.

Thanks and have a great day!

Regards,

Raúl

On Oct 6, 2023, at 12:25 PM, Raul Macias @.***> wrote:

 Zoltan-

I did, they usually take 24 hours to answer. I appreciate the effort. Thank you very much.

Raul

From: Zoltan Toth-Czifra @.> Sent: Friday, October 6, 2023 12:17 PM To: tcz/rotary-gpt @.> Cc: Raul Macias @.>; Mention @.> Subject: Re: [tcz/rotary-gpt] Audio Get Code off (Issue #2)

Seems like this did nothing. Would you mind sending my response to Grandstream to see what they say to this?

2 (comment)https://github.com/tcz/rotary-gpt/issues/2#issuecomment-1750773588

Maybe they will suggest using the same socket, I just don't really get why. Especially because it works well on the old hardware.

— Reply to this email directly, view it on GitHubhttps://github.com/tcz/rotary-gpt/issues/2#issuecomment-1751030988, or unsubscribehttps://github.com/notifications/unsubscribe-auth/A5645QFLKY6IFE4A7VAHCODX6AVIRAVCNFSM6AAAAAA4HWEL5CVHI2DSMVQWIX3LMV43OSLTON2WKQ3PNVWWK3TUHMYTONJRGAZTAOJYHA. You are receiving this because you were mentioned.Message ID: @.**@.>>