res_pjsip_session: outgoing call did not offer all configured codecs
For some scenarios when an outgoing call was made only a subset of the configured codecs were offered. If the codecs being offered happened to not have a codec supported by the phone then the call would fail. For instance Alice and Bob both are configured in Asterisk for g722 and ulaw( allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains g722 and not both g722 and ulaw, so the call ends. This patch makes it so all the audio codecs configured on the endpoint always get sent, and not just a subset. However priority is given to those codecs that are compatible with the "other side". ASTERISK-27259 #close Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696
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@ -2459,6 +2459,16 @@ struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint
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continue;
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continue;
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}
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}
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if (ast_stream_get_type(req_stream) == AST_MEDIA_TYPE_AUDIO) {
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/*
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* By appending codecs from the endpoint after compatible ones this
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* guarantees that priority is given to those while also allowing
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* translation to occur for non-compatible.
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*/
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ast_format_cap_append_from_cap(joint_cap,
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endpoint->media.codecs, AST_MEDIA_TYPE_AUDIO);
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}
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ast_stream_set_formats(clone_stream, joint_cap);
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ast_stream_set_formats(clone_stream, joint_cap);
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ao2_ref(joint_cap, -1);
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ao2_ref(joint_cap, -1);
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