SIP<>Airtel Outbound Calling

[SaurabhTiwari702]

I am using LiveKit SIP to make outbound calls via Airtel SIP trunk in India, but I am getting a 487 Request Terminated error. After debugging with Airtel, they confirmed the issue is related to missing user=phone parameter in SIP URIs.

Current Setup:

LiveKit SIP Server: 34.xxx.76
Airtel SIP Server: 117.xxx.5:5076
Transport: UDP
CLI Version: 2.12.9
SIP Image: livekit/sip:latest (pulled Jan 15, 2026)
The Issue:

Airtel requires user=phone parameter in the following SIP headers:
┌─────────────┬───────────────────────────────────────────────┬────────────────────────────────────────────┐
│ Header │ What LiveKit Sends │ What Airtel Requires │
├─────────────┼───────────────────────────────────────────────┼────────────────────────────────────────────┤
│ Request-URI │ sip:+91xxx@117.xxx.175:5076;transport=udp │ sip:+91xxx@117.96.31.175:5076;user=phone │
├─────────────┼───────────────────────────────────────────────┼────────────────────────────────────────────┤
│ From │ sip:+91xxx@34.xxx.76:5060;transport=udp │ sip:+91xxx@34.100.135.76:5060;user=phone │
├─────────────┼───────────────────────────────────────────────┼────────────────────────────────────────────┤
│ To │ sip:+91xxx@117.xxx.175:5076;transport=udp │ sip:+91xxx@117.xxx.175:5076;user=phone │
└─────────────┴───────────────────────────────────────────────┴────────────────────────────────────────────┘
What I Tried:

Added user=phone to custom headers (P-Asserted-Identity) - Works
Tried adding ;user=phone to trunk address field:
{“address”: “1XX.XX.XX.175:5076;user=phone”}
Error: trunk address should be a hostname or IP, not SIP URI
Tried with sip: prefix and template:
{“address”: “sip:{number}@1XX.XX.XX5:5076;user=phone”}
Error: trunk address should be a hostname or IP, not SIP URI
Call Flow (from PCAP):

LiveKit → INVITE → Airtel
Airtel → 100 Trying
Airtel → 183 Session Progress
Airtel → 487 Request Terminated
The call reaches Airtel and starts processing, but gets terminated due to missing user=phone in URIs.

My Question:

Is there a way to add user=phone parameter to the Request-URI, From, and To headers in outbound SIP calls?

Or is this a feature that needs to be added to LiveKit SIP?

[IshanSingla]

@SaurabhTiwari702
This is a known compatibility pain point with Indian PSTN carriers like Airtel, Jio, and BSNL — they follow stricter E.164 + user=phone enforcement compared to international SIP providers.
One practical workaround while this gets natively supported: put Asterisk or Kamailio as a middleware layer between LiveKit SIP and Airtel. LiveKit sends standard SIP to Asterisk/Kamailio, and you configure the outbound rewrite rule there to inject user=phone into the Request-URI, From, and To headers before it hits Airtel's trunk.
In Asterisk dialplan this is straightforward with SIPAddHeader or a PJSIP endpoint outbound header manipulation. Kamailio can do it with a simple append_hf or URI parameter rewrite in the request route.
Not ideal since it adds an extra hop, but it unblocks Airtel/similar Indian carriers today without waiting for a LiveKit fix. I've seen similar setups used with Tata Communications and BSNL SIP trunks for the same reason.
Longer term, a uri_params field on SIPOutboundTrunkInfo that appends custom parameters to the Request-URI would cleanly solve this for all carriers with similar requirements.

[murugancmi]

@SaurabhTiwari702 Hope you fix that config