res_pjsip: Redirect module and handling for MESSAGEs #1580

maximilianfridrich · 2025-11-07T11:35:57Z

res_pjsip: Introduce redirect module for handling 3xx responses
This commit introduces a new redirect handling module that provides
infrastructure for following SIP 3xx redirect responses. The redirect
functionality respects the endpoint's redirect_method setting and only
follows redirects when set to 'uri_pjsip'. This infrastructure can be
used by any PJSIP module that needs to handle 3xx redirect responses.

res_pjsip_messaging: Add support for following 3xx redirects
This commit integrates the redirect module into res_pjsip_messaging
to enable following 3xx redirect responses for outgoing SIP MESSAGEs.

When follow_redirect_methods contains 'message' on an endpoint, Asterisk
will now follow 3xx redirect responses for MESSAGEs, similar to how
it behaves for INVITE responses.

Resolves: #1576

UserNote: A new pjsip endpoint option follow_redirect_methods was added.
This option is a comma-delimited, case-insensitive list of SIP methods
for which SIP 3XX redirect responses are followed. An alembic upgrade
script has been added for adding this new option to the Asterisk
database.

maximilianfridrich · 2025-11-07T11:36:17Z

cherry-pick-to: 20
cherry-pick-to: 22
cherry-pick-to: 23
multiple-commits: standalone
testsuite-test-pr: 117

github-actions · 2025-11-07T12:00:57Z

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github-actions · 2025-11-09T13:49:05Z

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github-actions · 2025-11-18T12:22:17Z

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jcolp · 2025-11-25T09:58:26Z

include/asterisk/res_pjsip_redirect.h

+/*!
+ * \brief Maximum size for a URI string
+ */
+#define AST_SIP_MAX_URI_SIZE 512


Is there a reason not to use the PJSIP defined one?

I thought there was a PJSIP defined one but I couldn't find it. I found it now and will use it. Thanks.

include/asterisk/res_pjsip_redirect.h

jcolp · 2025-11-25T09:59:55Z

include/asterisk/res_pjsip_redirect.h

+ * SIP 3xx redirects, including visited URIs for loop detection,
+ * pending contacts for retry logic, and hop counting.
+ */
+struct ast_sip_redirect_context;


Context has a rather specific meaning within Asterisk, and I must admit on reading this name I thought it meant dialplan context and was confused until the brief description sunk in. I think a different name would eliminate that confusion. "ast_sip_redirect_state"

Good point, thanks and sorry about the confusion. I'll rename everything to state.

jcolp · 2025-11-25T10:00:42Z

include/asterisk/res_pjsip_redirect.h

+ * \brief Check if redirect should be followed based on endpoint configuration
+ *
+ * \param endpoint The SIP endpoint
+ * \param status_code The response status code


The response status code from what?

I clarified the doc comment: "The status code from a SIP response message (e.g. 305)"

res/res_pjsip/redirect.c

jcolp · 2025-11-25T10:11:08Z

res/res_pjsip/redirect.c

+ * \param contacts List to populate with parsed contacts
+ * \return Number of valid contacts found
+ */
+static int parse_redirect_contacts(pjsip_rx_data *rdata, struct redirect_contact_list *contacts)


Pass in the state/context pointer, and use the endpoint to add name to the logging

I'm kind of a broken record lately with logging, but giving information on which endpoint is involved can be very valuable if someone goes back to try to understand what happened on a well used system

The same applies for the code going forward after this comment, I'm not going to comment on each one but I urge you to go back to each and try to make them more specific/detailed including endpoint name

Understood. I tried to add the endpoint ID to the logs (mostly within the prose). Let me know if this is okay or if you would like it differently (e.g. a consistent format).

jcolp · 2025-11-25T10:13:07Z

res/res_pjsip/redirect.c

+		return -1;
+	}
+
+	ast_log(LOG_NOTICE, "Redirect: found %d Contact header%s\n", contact_count, contact_count == 1 ? "" : "s");


This should be removed

jcolp · 2025-11-25T10:14:45Z

res/res_pjsip_messaging.c

+{
+	struct message_response_data *resp_data;
+
+	resp_data = ao2_alloc(sizeof(*resp_data), message_response_data_destroy);


This can be allocated without a lock using ao2_alloc_options

Thanks. Done.

jcolp · 2025-11-25T10:15:56Z

res/res_pjsip_messaging.c

 	struct msg_data *mdata = data; /* The caller holds a reference */
+	/* Callback data for redirect handling */
+	struct message_response_data *resp_data;
+	char content_type_buf[128];


Where did 128 come from?

Arbitrary. I thought that should be enough for the Content-Type header which contains a MIME type (potentially with parameters). I changed it now to a malloced ast_str which can grow if needed.

maximilianfridrich · 2025-11-25T13:27:51Z

Thanks for the review @jcolp! I've incorporated your feedback and responded to your comments. Let me know what you think.

github-actions · 2025-11-25T13:47:23Z

Workflow PRCheck completed successfully

jcolp · 2025-12-08T09:49:09Z

res/res_pjsip/pjsip_config.xml

 						and also supporting multiple potential redirect targets. The con is that since redirection occurs
 						within chan_pjsip redirecting information is not forwarded and redirection can not be
-						prevented.
+						prevented. Further, if <literal>uri_pjsip</literal> is configured, 3xx responses to SIP MESSAGEs


I think the follow_redirect_methods is good.

jcolp · 2025-12-08T09:51:39Z

res/res_pjsip_messaging.c

+
+	/* Check event type */
+	if (e->body.tsx_state.type == PJSIP_EVENT_TIMER) {
+		ast_debug(1, "MESSAGE request timed out\n");


Needs to include endpoint if present and be more specific "MESSAGE request timed out on sending to endpoint %s\n"

This commit introduces a new redirect handling module that provides infrastructure for following SIP 3xx redirect responses. The redirect functionality respects the endpoint's redirect_method setting and only follows redirects when set to 'uri_pjsip'. This infrastructure can be used by any PJSIP module that needs to handle 3xx redirect responses.

maximilianfridrich · 2025-12-12T11:13:40Z

Thanks again for another review @jcolp! I've incorporated your feedback, created the new config option follow_redirect_methods with corresponding alembic migration and adapted the testsuite PR accordingly to use this new option.

github-actions

Attention! This pull request may contain issues that could prevent it from being accepted. Please review the checklist below and take the recommended action. If you believe any of these are not applicable, just add a comment and let us know.

An Alembic change was detected but a commit message UpgradeNote with at least one of the 'alembic', 'database' or 'schema' keywords wasn't found. Please add an UpgradeNote to the commit message that mentions one of those keywords notifying users that there's a database schema change.

Documentation:

Asterisk Developer Documentation

github-actions · 2025-12-12T11:21:20Z

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This commit integrates the redirect module into res_pjsip_messaging to enable following 3xx redirect responses for outgoing SIP MESSAGEs. When follow_redirect_methods contains 'message' on an endpoint, Asterisk will now follow 3xx redirect responses for MESSAGEs, similar to how it behaves for INVITE responses. Resolves: asterisk#1576 UserNote: A new pjsip endpoint option follow_redirect_methods was added. This option is a comma-delimited, case-insensitive list of SIP methods for which SIP 3XX redirect responses are followed. An alembic upgrade script has been added for adding this new option to the Asterisk database.

github-actions · 2025-12-12T12:14:06Z

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master-pjs5: FAILED TEST: channels/pjsip/subscriptions/rls/lists_of_lists/nominal/mwi/batched

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channels/pjsip: Add tests for SIP MESSAGE redirect functionality asterisk/testsuite#117

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maximilianfridrich force-pushed the pjsip_redirect_module branch from 683e974 to 788d3d7 Compare November 9, 2025 13:21

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res_pjsip: Redirect module and handling for MESSAGEs #1580

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res_pjsip: Redirect module and handling for MESSAGEs #1580

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