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chan_pjsip: Disable SSRC change for WebRTC endpoints.

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Commit b333ee3b introduced a fix to chan_pjsip that addressed RTP issues with
blind transfers and some SBCs.  Unfortunately, the fix broke some WebRTC
clients that are sensitive to SSRC changes and non-monotonic timestamps so
the fix is now disabled for endpoints with the "bundle" parameter set to true.

Resolves: #1535
This commit is contained in:
George Joseph
2025-10-21 09:34:23 -06:00
parent 0019b27f5c
commit 8e8a0a1e82
1 changed files with 8 additions and 6 deletions
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14
channels/chan_pjsip.c
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@@ -1820,12 +1820,14 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi
case AST_CONTROL_SRCUPDATE:
break;
case AST_CONTROL_SRCCHANGE:
/* Generate a new SSRC due to media source change and RTP timestamp reset.
Ensures RFC 3550 compliance and avoids SBC interoperability issues (Sonus/Ribbon)*/
for (i = 0; i < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions); ++i) {
media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, i);
if (media && media->rtp) {
ast_rtp_instance_change_source(media->rtp);
if (!channel->session->endpoint->media.bundle) {
/* Generate a new SSRC due to media source change and RTP timestamp reset.
Ensures RFC 3550 compliance and avoids SBC interoperability issues (Sonus/Ribbon)*/
for (i = 0; i < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions); ++i) {
media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, i);
if (media && media->rtp) {
ast_rtp_instance_change_source(media->rtp);
}
}
}
break;