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	chan_rtp: Use μ-law by default instead of signed linear
Multicast/Unicast RTP do not use SDP so we need to use a format that cleanly maps to one of the static RTP payload types. Without this change, an Originate to a Multicast or Unicast channel without a format specified would produce no audio on the receiving device. ASTERISK-21399 #close Reported by: Tzafrir Cohen Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3
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		| @@ -117,6 +117,22 @@ static int rtp_hangup(struct ast_channel *ast) | ||||
| 	return 0; | ||||
| } | ||||
|  | ||||
| static struct ast_format *derive_format_from_cap(struct ast_format_cap *cap) | ||||
| { | ||||
| 	struct ast_format *fmt = ast_format_cap_get_format(cap, 0); | ||||
|  | ||||
| 	if (ast_format_cap_count(cap) == 1 && fmt == ast_format_slin) { | ||||
| 		/* | ||||
| 		 * Because we have no SDP, we must use one of the static RTP payload | ||||
| 		 * assignments. Signed linear @ 8kHz does not map, so if that is our | ||||
| 		 * only capability, we force μ-law instead. | ||||
| 		 */ | ||||
| 		fmt = ast_format_ulaw; | ||||
| 	} | ||||
|  | ||||
| 	return fmt; | ||||
| } | ||||
|  | ||||
| /*! \brief Function called when we should prepare to call the multicast destination */ | ||||
| static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause) | ||||
| { | ||||
| @@ -171,7 +187,7 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo | ||||
|  | ||||
| 	fmt = ast_multicast_rtp_options_get_format(mcast_options); | ||||
| 	if (!fmt) { | ||||
| 		fmt = ast_format_cap_get_format(cap, 0); | ||||
| 		fmt = derive_format_from_cap(cap); | ||||
| 	} | ||||
| 	if (!fmt) { | ||||
| 		ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n", | ||||
| @@ -298,7 +314,7 @@ static struct ast_channel *unicast_rtp_request(const char *type, struct ast_form | ||||
| 			goto failure; | ||||
| 		} | ||||
| 	} else { | ||||
| 		fmt = ast_format_cap_get_format(cap, 0); | ||||
| 		fmt = derive_format_from_cap(cap); | ||||
| 		if (!fmt) { | ||||
| 			ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n", | ||||
| 				args.destination); | ||||
|   | ||||
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