Invalid SDP offer in starting SIP call

andrea.mason85 · November 14, 2025, 9:25am

Hi all,

Since this morning, we’ve been facing a problem when starting a SIP call to OpenAI. It seems that the SDP cannot be established correctly. Everything started around 8 AM GMT.

We have an Asterisk server that receives calls from a provider and forwards them to OpenAI.

Here is an example of a SIP call that generates this error, captured on Asterisk:


Reliably Transmitting (NAT) to 1XX.6X.XXX.XXX:5061:
INVITE sip:proj_axxxxxxxxxxxxx@sip.api.openai.com:5061 SIP/2.0
Via: SIP/2.0/TLS 1X.1XX.XXX.XXX:5061;branch=z9hG4bK0afaee5b;rport
Max-Forwards: 70
From: "+393XXXXXXXXX" <sip:+393XXXXXXXXX@1X.1XX.XXX.XXX>;tag=as021b85bc
To: <sip:proj_axxxxxxxxxxxxx@sip.api.openai.com:5061>
Contact: <sip:+393XXXXXXXXX@1X.1XX.XXX.XXX:5061;transport=tls>
Call-ID: 66f41e1313fc71677075a6c56138f2f3@1X.1XX.XXX.XXX:5061
CSeq: 102 INVITE
User-Agent: ucontact
Date: Fri, 14 Nov 2025 08:59:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 338

v=0
o=root 1770862544 1770862544 IN IP4 1X.1XX.XXX.XXX
s=Asterisk PBX 13.38.1
c=IN IP4 1X.1XX.XXX.XXX
t=0 0
m=audio 12864 RTP/AVP 8 0 18 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/OPENAI_SIP_TRUNK/proj_a5CSQHrkYrzOjqagkibJtmxj/

<--- SIP read from TLS:1XX.6X.XXX.XXX:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 1X.1XX.XXX.XXX:5061;received=188.114.102.87;branch=z9hG4bK0afaee5b;rport=33826
From: "+393XXXXXXXXX" <sip:+393XXXXXXXXX@1X.1XX.XXX.XXX>;tag=as021b85bc
To: <sip:proj_axxxxxxxxxxxxx@sip.api.openai.com:5061>;tag=70f6ba52-362d-4075-bd7e-5fb1454d033a
Call-ID: 66f41e1313fc71677075a6c56138f2f3@1X.1XX.XXX.XXX:5061
CSeq: 102 INVITE
Content-Length: 0
Contact: <sips:OAI@sip.api.openai.com:5061>

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from TLS:1XX.6X.XXX.XXX:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 1X.1XX.XXX.XXX:5061;rport=33826;branch=z9hG4bK0afaee5b;received=188.114.102.87
From: "+393XXXXXXXXX" <sip:+393XXXXXXXXX@1X.1XX.XXX.XXX>;tag=as021b85bc
To: <sip:proj_axxxxxxxxxxxxx@sip.api.openai.com:5061>;tag=70f6ba52-362d-4075-bd7e-5fb1454d033a
Call-ID: 66f41e1313fc71677075a6c56138f2f3@1X.1XX.XXX.XXX:5061
CSeq: 102 INVITE
Content-Length: 0
Contact: <sips:OAI@sip.api.openai.com:5061>

<------------->
--- (8 headers 0 lines) ---
sip_route_dump: route/path hop: <sips:OAI@sip.api.openai.com:5061>
    -- SIP/OPENAI_SIP_TRUNK-0000003c is ringing
Reliably Transmitting (NAT) to 127.0.0.1:5060:
OPTIONS sip:127.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 1X.1XX.XXX.XXX:5060;branch=z9hG4bK25e6c229;rport
Max-Forwards: 70
From: "uContact" <sip:uContact@1X.1XX.XXX.XXX>;tag=as1b07c624
To: <sip:127.0.0.1>
Contact: <sip:uContact@1X.1XX.XXX.XXX:5060>
Call-ID: 2faae62d1cef72272b91ae715da63dc3@1X.1XX.XXX.XXX:5060
CSeq: 102 OPTIONS
User-Agent: ucontact
Date: Fri, 14 Nov 2025 08:59:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0




<--- SIP read from TLS:1XX.6X.XXX.XXX:5061 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/TLS 1X.1XX.XXX.XXX:5061;branch=z9hG4bK0afaee5b;rport=33826;received=188.114.102.87
From: "+393XXXXXXXXX" <sip:+393XXXXXXXXX@1X.1XX.XXX.XXX>;tag=as021b85bc
To: <sip:proj_a5CSQHrkYrzOjqagkibJtmxj@sip.api.openai.com:5061>;tag=70f6ba52-362d-4075-bd7e-5fb1454d033a
Call-ID: 66f41e1313fc71677075a6c56138f2f3@1X.1XX.XXX.XXX:5061
CSeq: 102 INVITE
Content-Length: 159
Content-Type: application/json
Contact: <sips:OAI@sip.api.openai.com:5061>

{
    "error": {
        "message": "Invalid SDP offer.",
        "type": "invalid_request_error",
        "code": "invalid_offer",
        "param": ""
    }
}
<------------->
--- (9 headers 8 lines) ---
Transmitting (NAT) to 1XX.6X.XXX.XXX:5061:
ACK sips:OAI@sip.api.openai.com:5061 SIP/2.0
Via: SIP/2.0/TLS 1X.1XX.XXX.XXX:5061;branch=z9hG4bK0afaee5b;rport
Max-Forwards: 70
From: "+393XXXXXXXXX" <sip:+393XXXXXXXXX@1X.1XX.XXX.XXX>;tag=as021b85bc
To: <sip:proj_a5CSQHrkYrzOjqagkibJtmxj@sip.api.openai.com:5061>;tag=70f6ba52-362d-4075-bd7e-5fb1454d033a
Contact: <sip:+393XXXXXXXXX@1X.1XX.XXX.XXX:5061;transport=tls>
Call-ID: 66f41e1313fc71677075a6c56138f2f3@1X.1XX.XXX.XXX:5061
CSeq: 102 ACK
User-Agent: ucontact
Content-Length: 0


---

Has anyone else experienced the same issue this morning?

Thanks

s-nishihara1 · November 14, 2025, 9:51am

I’m also experiencing issues with the Realtime API’s SIP connection starting from around 8:00 a.m. Japan time.

The symptom is that the model’s utterance suddenly cuts off in the middle. However, the WebSocket remains connected.

We’re using Google Cloud Run as a tunneling tool, and nothing appears to be wrong in its logs, but it seems that the Realtime API is unilaterally stopping the conversation.

andrea.mason85 · November 14, 2025, 10:03am

Same as we’re getting in this moment s-nishihara1

In order to get the complete situation, we tried to disable Alawn in SDP, and now calls starts but, after more or less a minute, is cutted off and the wss, even if it’s up, is not receiving any event

Ivan_Klyuchnikov · November 14, 2025, 10:05am

Hi, @andrea.mason85

I can confirm the same issue

SIP/2.0 400 Bad Request
Contact: sips:OAI@sip.api.openai.com:5061

{
“error”: {
“message”: “Invalid SDP offer.”,
“type”: “invalid_request_error”,
“code”: “invalid_offer”,
“param”: “”
}
}

s-nishihara1 · November 14, 2025, 10:18am

We are making full use of the Realtime API for handling customer support at hotels in Japan. For now, we’ve built a Twilio flow that connects customers to human agents instead of the Realtime API, but we hope this issue will be resolved as soon as possible.

dw.arinaudo · November 14, 2025, 10:35am

same here, we are offline as of today.

philip · November 14, 2025, 11:59am

This issue also appears to reproduce using the Agents SDK Twilio SIP example with Twilio as the SIP provider.

Calls connect as expected and the agent is initially responsive, but the realtime audio is cut off mid-sentence after 30 seconds to 1 minute.

The call remains connected until hung up by the caller, but no further audio is heard.

Enhancing the logging in the example didn’t reveal any specific errors. Transcription of the agent’s speech stops a few words after the last audible utterance. No event indicating the turn has completed is observed.

andrea.mason85 · November 14, 2025, 12:05pm

Misery loves company

I’ve opened a ticket , Hope It Will be solved shortly

magnus.lofqvist · November 14, 2025, 12:23pm

Hi,

I’m experiencing the same issue. First, alaw stops working with an “invalid SDP offer” error, and when using ulaw it stops functioning after 30–60 seconds.

Best regards,
Magnus

Gustavo_Lima · November 14, 2025, 1:45pm

same here, our agent is completely offline

andrea.mason85 · November 14, 2025, 1:56pm

So problem seems spread all around the world (at least Italy and Japan).
@OpenAI_Support seems a big problem…

Gustavo_Lima · November 14, 2025, 1:58pm

we’re located in south america, so yeah… seems a big problem

madrez · November 14, 2025, 2:02pm

Same here in the UK and it is affecting our customer deliveries…

andrea.mason85 · November 14, 2025, 2:40pm

@OpenAIAPIhelper @OpenAI_Support @Sean-Der Please check it, seems all around the world SIP isn’t working at all

spido · November 14, 2025, 2:55pm

Same here in Belgium.

As far as i know, they changed something this morning in the SIP behaviour.
In the past we could pass multiple codecs in the SDP offers.
Now the SIP endpoint refuse in a Bad Request reply all not allowed codecs.

So to prevent that we need to pass ONLY the PCMU (ulaw) codecs in the SDP.
The call works with this change but can only last 40/50 seconds.

I do some packet capture to analyze the RTP trafic.

The RTP session is well keeped and still alive all along. But suddenly the RTP packets received (sent from openai) become completely blank as we can see on the capture.

websocket is still alive also.

@OpenAI_Support for the record and the analysis.

Sean-Der · November 14, 2025, 2:56pm

Thank you so much for the ping! I am able to reproduce this on 1800-ChatGPT also.

Investigating now

spido · November 14, 2025, 3:04pm

This could be great to report this in a “SIP” status category on your https://status.openai.com/

As we resell products based on your stack, we could redirect the end customers to see by their own that something is broken there.

Thanks for your investigation and further solution

andrea.mason85 · November 14, 2025, 3:05pm

Yes @spido it could be very useful having a specific section in status in order to check it.

Gustavo_Lima · November 14, 2025, 3:09pm

100% agree on this! even though we should implement proper ways to rollback if a call fails, we should be able to monitor status of the sip connector

Sean-Der · November 14, 2025, 3:13pm

Rollback is ~50% done now. should be fully recovered in ~10 more mins.

I will work on getting a dedicated status page up and getting a test that ensures audio works for longer calls.

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Invalid SDP offer in starting SIP call

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