Re: [EXTERNAL] Webrtc->SIP Audio Issues

Richard Fuchs via sr-users <[email protected]>
Newsgroups gmane.comp.voip.ser
Organization Sipwise GmbH
Message-ID <[email protected]>
On 18/09/2025 13.57, Fernando Lopes via sr-users wrote:
> My logs:
> ...
> [1758217893.109379] INFO: [ID="m7e648isjm9n96ti62k9"]: [control] Replying to 'answer' from 10.5.0.8:42938 (elapsed time 0.000804 sec)
> [1758217893.109411] INFO: [ID="*m7e648isjm9n96ti62k9" tag="f03487ad-0be2-49d5-9452-289c407db828"* index="1" port="14656"]: [ice] ICE negotiated: peer for component 1 is 10.5.0.2:28384
> ...
>
> [1758217930.743494] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] Closing call due to timeout
> [1758217930.743636] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] Final packet stats:
> [1758217930.743651] INFO: [ID=*"483961fa065a4df9b5f76059c8a1a3e3"*]: [core] --- Tag '*2bad2cacef0945b1a506a198aeb4ae01'*, created 1:00 ago for branch ''
> ...
> [1758217930.743683] INFO: [ID="483961fa065a4df9b5f76059c8a1a3e3"]: [core] --- Tag*'88cf8c49-cdfc-475e-8075-a3ce0a36ecca'*, created 1:00 ago for branch ''
Are you sure these belong to the same call? With the call IDs and tags 
being different and all...
> My Kamailio code:
> route[GET_ICE] {
>      if ( $proto == "wss" ) {
>          if (is_ip_rfc1918("$si")) {
>              $dlg_var(ice) = " ";
>          } else {
>              $dlg_var(ice) = " ICE=remove RTP ";
>
>          }
>      } else {
>          $dlg_var(ice) = "ICE=FORCE";
>      }
> }

You mention WebRTC <> Asterisk, but I assume you really want to bridge 
it to plain SIP/RTP? Making that work requires more than just making ICE 
work on one side.

WebRTC requires a specific set of features, and so offers *going to* 
WebRTC should have the following options set:

transport-protocol=UDP/TLS/RTP/SAVPF ICE=force trickle-ICE rtcp-mux-offer rtcp-mux-require no-rtcp-attribute SDES=off generate-mid

If your version of rtpengine is recent enough, a convenience flag 
`WebRTC` is provided that enables all of these in one go.

Offers *going to* plain SIP/RTP generally require removal of these 
features, depending on how picky the intended recipient is and what 
features it supports. Generally you want to remove SRTP encryption, ICE, 
and RTCP multiplexing:

transport-protocol=RTP/AVP ICE=remove rtcp-mux-demux

Subsequent offers (re-invites) and answers (200 OK etc) may omit any or 
all of these.

Good luck

Cheers

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