[Asterisk-code-review] chan_sip: Allow [peer] without audio (text+video). (asterisk[18])

[George Joseph]

George Joseph has submitted this change. ( https://gerrit.asterisk.org/c/asterisk/+/15410 )

Change subject: chan_sip: Allow [peer] without audio (text+video).
......................................................................

chan_sip: Allow [peer] without audio (text+video).

Two previous commits, 620d9f4 and 6d980de, allow to set up a call
without audio, again. That was introduced originally with commit f04d5fb
but changed and broke over time. The original commit missed one
scenario: A [peer] section in sip.conf, which does not allow audio at
all. In that case, chan_sip rejected the call, although even when the
requester offered no audio. Now, chan_sip does not check whether there
is no audio format but checks whether there is no format in general. In
other words, if there is at least one format to offer, the call succeeds.

However, to prevent calls with no-audio, chan_sip still rejects calls
when both call parties (caller = requester of the call *and* callee =
[peer] section in sip.conf) included audio. In such a case, it is
expected that the call should have audio.

ASTERISK-29280

Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e
---
M channels/chan_sip.c
1 file changed, 15 insertions(+), 2 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  Kevin Harwell: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved; Approved for Submit



diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index e5d6cb6..47db9e5 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6555,8 +6555,21 @@
 	ast_rtp_instance_available_formats(p->rtp, p->caps, p->prefcaps, p->jointcaps);
 	p->jointnoncodeccapability = p->noncodeccapability;
 
-	/* If there are no audio formats left to offer, punt */
-	if (!(ast_format_cap_has_type(p->jointcaps, AST_MEDIA_TYPE_AUDIO))) {
+	/* If there are no formats left to offer, punt */
+	if (ast_format_cap_empty(p->jointcaps)) {
+		ast_log(LOG_WARNING, "No format found to offer. Cancelling call to %s\n", p->username);
+		res = -1;
+	/* If audio was requested (prefcaps) and the [peer] section contains
+	 * audio (caps) the user expects audio. In that case, if jointcaps
+	 * contain no audio, punt. Furthermore, this check allows the [peer]
+	 * section to have no audio. In that case, the user expects no audio
+	 * and we can pass. Finally, this check allows the requester not to
+	 * offer any audio. In that case, the call is expected to have no audio
+	 * and we can pass, as well.
+	 */
+	} else if ((ast_format_cap_empty(p->caps) || ast_format_cap_has_type(p->caps, AST_MEDIA_TYPE_AUDIO)) &&
+		   (ast_format_cap_empty(p->prefcaps) || ast_format_cap_has_type(p->prefcaps, AST_MEDIA_TYPE_AUDIO)) &&
+		   !ast_format_cap_has_type(p->jointcaps, AST_MEDIA_TYPE_AUDIO)) {
 		ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
 		res = -1;
 	} else {

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Gerrit-Project: asterisk
Gerrit-Branch: 18
Gerrit-Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e
Gerrit-Change-Number: 15410
Gerrit-PatchSet: 2
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at sangoma.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-MessageType: merged
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