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sip: media: Allow specifying SRTP for GStreamer capabilities

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When using SRTP the GstCaps must be set accordingly.
This commit is contained in:
Evangelos Ribeiro Tzaras
2022-04-07 11:42:18 +02:00
parent 7ac862155b
commit 58f9f5cb62
3 changed files with 8 additions and 4 deletions
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7
plugins/sip/gst-rfc3551.c
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@@ -106,15 +106,18 @@ media_codec_by_payload_id (guint payload_id)
/* media_codec_get_gst_capabilities:
*
* @codec: A #MediaCodecInfo
* @use_srtp: Whether to use SRTP
*
* Returns: (transfer full): The capability string describing GstCaps.
* Used for the RTP source element.
*/
gchar *
media_codec_get_gst_capabilities (MediaCodecInfo *codec)
media_codec_get_gst_capabilities (MediaCodecInfo *codec,
gboolean use_srtp)
{
return g_strdup_printf ("application/x-rtp,media=(string)audio,clock-rate=(int)%u"
return g_strdup_printf ("application/%s,media=(string)audio,clock-rate=(int)%u"
",encoding-name=(string)%s,payload=(int)%u",
use_srtp ? "x-srtp" : "x-rtp",
codec->clock_rate,
codec->name,
codec->payload_id);