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 19/09/2025 07.25, Fernando Lopes via sr-users wrote: > [core] --- Tag '7eiernq2ig', created 1:51 ago for branch '' > [core] ------ Media #1*(audio over UDP/TLS/RTP/SAVPF) using unknown codec* > [core] --------- Port 10.5.0.8:14934 <> 176.223.17.217:55146, SSRC 0, in 0 p, 0 b,*456 e, *111 ts, out 0 p, 0 b, 0 e > [core] --- Tag 'c73ccc40-a836-4ac7-8968-0d84f7d40b67', created 1:51 ago for branch '' > [core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using G722/8000 > [core] --------- Port 10.5.0.8:11352 <> 10.5.0.2:22438, SSRC 221a786, in 456 p, 41900 b, 0 e, 98 ts, out 7 p, 1946 b, 0 e This says you were running DTLS-SRTP on both sides of the call. Probably not what you wanted, since the first side is only reporting errors and no valid audio received. Check your signalling to rtpengine, probably you're using the same flags for both offers and answers, or possibly no flags at all. Enable debug logging in rtpengine to see exactly what's coming through. Cheers __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- [email protected] To unsubscribe send an email to [email protected] Important: keep the mailing list in the recipients, do not reply only to the sender!