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chan_pjsip.c: Change SSRC after media source change

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When the RTP media source changes, such as after a blind transfer, the new source introduces a discontinuous timestamp. According to RFC 3550, Section 5.1, an RTP stream's timestamp for a given SSRC must increment monotonically and linearly.
To comply with the standard and avoid a large timestamp jump on the existing SSRC, a new SSRC is generated for the new media stream.
This change resolves known interoperability issues with certain SBCs (like Sonus/Ribbon) that stop forwarding media when they detect such a timestamp violation. This code uses the existing implementation from chan_sip.

Resolves: #927
This commit is contained in:
Martin Tomec
2025-06-25 15:20:17 +02:00
committed by github-actions[bot]
parent 415daae95f
commit 9e7788ad21
1 changed files with 8 additions and 0 deletions
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8
channels/chan_pjsip.c
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@@ -1820,6 +1820,14 @@ static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const voi
case AST_CONTROL_SRCUPDATE: case AST_CONTROL_SRCUPDATE:
break; break;
case AST_CONTROL_SRCCHANGE: 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);
}
}
break; break;
case AST_CONTROL_REDIRECTING: case AST_CONTROL_REDIRECTING:
if (ast_channel_state(ast) != AST_STATE_UP) { if (ast_channel_state(ast) != AST_STATE_UP) {