Anyone successfully integrated livekit sip with Avaya

[PrakashSuthar]

I have been trying to integrate Avaya with LiveKit SIP server to transfer some existing IVR calls to our LiveKit SIP server. We were able to establish direct connection to SIP server, where we verified audio is received.
However, when I transfer the call in the middle of ivr flow using a vdn configured on Avaya, the livekit throws 500 error with no audio in sdp.

Error logs:

2025-11-06T12:29:39.612Z	DEBUG	Getting connection	{"nodeID": ".....", "caller": "transport<TCP>", "addr": "ip1:port1"}
2025-11-06T12:29:39.612Z	INFO	sip	sip/inbound.go:178	processing invite	{"nodeID": ".....", "callID": "<callId>", "fromIP": "ip1", "toIP": "mydc:5060", "fromHost": "mydc", "fromUser": "myPhoneNumber", "toHost": "mydc", "toUser": "<vdn>", "sipTag": "....", "sipCallID": "<sipcallId>"}
2025-11-06T12:29:39.615Z	DEBUG	sip	sip/inbound.go:588	SDP offer	{"nodeID": ".....", "callID": "<callId>", "fromIP": "ip1", "toIP": "mydc:5060", "fromHost": "mydc", "fromUser": "myPhoneNumber", "toHost": "mydc", "toUser": "<vdn>", "sipTag": "....", "sipCallID": "<sipcallId>", "sipTrunk": "<sipTrunkId>", "jitterBuf": false, "sipTrunk": "<sipTrunkId>", "sipRule": "<sipRule>", "sdp": ""}
2025-11-06T12:29:39.615Z	DEBUG	sip	sip/media_port.go:158	listening for media on UDP	{"nodeID": ".....", "callID": "<callId>", "fromIP": "ip1", "toIP": "mydc:5060", "fromHost": "mydc", "fromUser": "myPhoneNumber", "toHost": "mydc", "toUser": "<vdn>", "sipTag": "....", "sipCallID": "<sipcallId>", "sipTrunk": "<sipTrunkId>", "jitterBuf": false, "sipTrunk": "<sipTrunkId>", "sipRule": "<sipRule>", "port": port2}
2025-11-06T12:29:39.615Z	ERROR	sip	sip/inbound.go:462	Cannot start media	{"nodeID": ".....", "callID": "<callId>", "fromIP": "ip1", "toIP": "mydc:5060", "fromHost": "mydc", "fromUser": "myPhoneNumber", "toHost": "mydc", "toUser": "<vdn>", "sipTag": "....", "sipCallID": "<sipcallId>", "sipTrunk": "<sipTrunkId>", "jitterBuf": false, "sipTrunk": "<sipTrunkId>", "sipRule": "<sipRule>", "error": "no audio in sdp"}
github.com/livekit/sip/pkg/sip.(*inboundCall).handleInvite.func2
	/workspace/pkg/sip/inbound.go:462
github.com/livekit/sip/pkg/sip.(*inboundCall).handleInvite
	/workspace/pkg/sip/inbound.go:515
github.com/livekit/sip/pkg/sip.(*Server).processInvite
	/workspace/pkg/sip/inbound.go:274
github.com/livekit/sip/pkg/sip.(*Server).onInvite
	/workspace/pkg/sip/inbound.go:128
github.com/livekit/sipgo.(*Server).handleRequest
	/go/pkg/mod/github.com/livekit/sipgo@v0.13.2-0.20250601220430-a77cc3f220fb/server.go:235
2025-11-06T12:29:39.615Z	DEBUG	Server transaction terminating	{"nodeID": ".....", "caller": "transaction.Layer"}
2025-11-06T12:29:39.615Z	DEBUG	Server transaction destroyed	{"nodeID": ".....", "caller": "transaction.Layer", "tx": "z9hG4bK-s1632-001176478504-1--s1632-__ip1__5060__INVITE"}
2025-11-06T12:29:39.616Z	WARN	sip	sip/inbound.go:780	Closing inbound call with error	{"nodeID": ".....", "callID": "<callId>", "fromIP": "ip1", "toIP": "mydc:5060", "fromHost": "mydc", "fromUser": "myPhoneNumber", "toHost": "mydc", "toUser": "<vdn>", "sipTag": "....", "sipCallID": "<sipcallId>", "sipTrunk": "<sipTrunkId>", "jitterBuf": false, "sipTrunk": "<sipTrunkId>", "sipRule": "<sipRule>", "status": 486, "reason": "media-failed"}
2025-11-06T12:29:39.616Z	INFO	sip	sip/inbound.go:801	Inbound call closed	{"nodeID": ".....", "callID": "<callId>", "fromIP": "ip1", "toIP": "mydc:5060", "fromHost": "mydc", "fromUser": "myPhoneNumber", "toHost": "mydc", "toUser": "<vdn>", "sipTag": "....", "sipCallID": "<sipcallId>", "sipTrunk": "<sipTrunkId>", "jitterBuf": false, "sipTrunk": "<sipTrunkId>", "sipRule": "<sipRule>", "status": 486, "reason": "media-failed"}
2025-11-06T12:29:39.616Z	INFO	sip	sip/inbound.go:777	call statistics	{"nodeID": ".....", "callID": "<callId>", "fromIP": "ip1", "toIP": "mydc:5060", "fromHost": "mydc", "fromUser": "myPhoneNumber", "toHost": "mydc", "toUser": "<vdn>", "sipTag": "....", "sipCallID": "<sipcallId>", "sipTrunk": "<sipTrunkId>", "jitterBuf": false, "sipTrunk": "<sipTrunkId>", "sipRule": "<sipRule>", "status": 486, "reason": "media-failed", "stats": {"port":{"streams":0,"packets":0,"packets_ignored":0,"packets_input":0,"mux_packets":0,"mux_bytes":0,"audio_packets":0,"audio_bytes":0,"dtmf_packets":0,"dtmf_bytes":0},"room":{"input_packets":0,"input_bytes":0,"mixer_samples":0,"mixer_frames":0,"output_samples":0,"output_frames":0},"mixer":{"tracks":0,"tracks_total":0,"restarts":0,"mixes":0,"mixes_timed":0,"mixes_jump":0,"mixes_zero":0,"input_samples":0,"input_frames":0,"mixed_samples":0,"mixed_frames":0,"output_samples":0,"output_frames":0}}}
2025-11-06T12:29:39.616Z	DEBUG	Getting connection	{"nodeID": ".....", "caller": "transport<TCP>", "addr": "ip1:port1"}
2025-11-06T12:29:39.616Z	DEBUG	Server transaction terminating	{"nodeID": ".....", "caller": "transaction.Layer"}
2025-11-06T12:29:39.616Z	DEBUG	Server transaction destroyed	{"nodeID": ".....", "caller": "transaction.Layer", "tx": "z9hG4bK-s1632-001176478504-1--s1632-__ip1__5060__INVITE"}
2025-11-06T12:29:39.616Z	DEBUG	Getting connection	{"nodeID": ".....", "caller": "transport<TCP>", "addr": "ip1:port1"}
2025-11-06T12:29:39.616Z	DEBUG	Server transaction terminating	{"nodeID": ".....", "caller": "transaction.Layer"}
2025-11-06T12:29:39.616Z	DEBUG	Server transaction destroyed	{"nodeID": ".....", "caller": "transaction.Layer", "tx": "z9hG4bK-s1632-000138059056-1--s1632-__ip1__5060__OPTIONS"}
2025-11-06T12:29:39.617Z	DEBUG	Server transaction terminating	{"nodeID": ".....", "caller": "transaction.Layer"}
2025-11-06T12:29:39.617Z	DEBUG	Server transaction destroyed	{"nodeID": ".....", "caller": "transaction.Layer", "tx": "z9hG4bK-s1632-001176478504-1--s1632-__ip1__5060__INVITE"}

[dennwc]

Right, this case assumes that the provider can send SDP in the middle of the call, which we do not support yet. There's some initial work happening on in here, but there's no ETA for this feature yet. Also related: #438.

[PrakashSuthar]

Mine was a blind transfer to LiveKit, hence, the transferrer system(Avaya) expects the LiveKit server to generate SIP INVITE with appropriate data in response to the SIP REFER.
Any other alternative I can achieve a mid call transfer?

[dennwc]

the transferrer system(Avaya) expects the LiveKit server to generate SIP INVITE

I'm pretty sure it's the other way around. There clearly an INVITE going into LiveKit. When SIP endpoints A and B establish a call, and A refers B elsewhere (C), then B sends invite to C. Since you are referring to LiveKit, some system must send INVITE to it. And this is exactly what's happening here. However, the INVITE doesn't contain an SDP and LiveKit doesn't support invites without SDP at the moment.