[asterisk-dev] [Code Review] 4442: chan_sip: Asterisk fails to re-activate an inactive media session when an offer does not contain a=sendrecv

Ashley Sanders reviewboard at asterisk.org
Thu Feb 26 11:35:02 CST 2015

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(Updated Feb. 26, 2015, 11:35 a.m.)

Review request for Asterisk Developers.


Applied the changes as suggested by Mark Michelson; made an adjustment (as suggested by file and mjordan on #asterisk-dev) to make the origin session id and version id different for re-invites.

(2015-02-26 09:45:12) mjordan: essentially, when you modify a session, you have to increase the version of the session you are modifying, otherwise we have to assume that it is a 'stale' (or repeated) offer for an existing session
(2015-02-26 09:45:26) file: which becomes a no-op
(2015-02-26 09:45:46) mjordan: so, in your INVITE request that sends an SDP that takes the call "off hold", the o= line should have the <version> number bumped by at least one

Bugs: ASTERISK-24824

Repository: testsuite


This test is to ensure that Asterisk correctly applies the direction of the media stream when a=<sendonly|recvonly|inactive|sendrecv> is missing from the offer's SDP. The expected behavior is for Asterisk to apply "sendrecv" as the direction of the media stream when no direction attribute is present in an offer's SDP. According to RFC 4566 (Section 6. SDP Attributes): "If none of the attributes "sendonly", "recvonly", "inactive", and "sendrecv" is present, "sendrecv" SHOULD be assumed as the default for sessions that are not of the conference type "broadcast" or "H332" [...]"

The test scenario:

1. From Phone A, send an offer to Phone B to establish a call
2. From Phone B, send an offer to Phone A to put the call on hold. 
3. Observe that the MOH start event occurs.
4. From Phone B, send an offer to Phone A to 'un-hold' the call (ensure that the direction attribute from the offer's SDP is omitted)
5. Observe that the MOH stop event occurs.

Presently, this test fails for certain versions of Asterisk. From what I can tell, it is present from (at least) 1.8.21 up to the 11 branch.

***Note*** This is the test. It is only the test. The update to the Asterisk source is coming soon to a review board near you (well, this review board).

Diffs (updated)

  ./asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_no_direction.xml PRE-CREATION 
  ./asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml 6458 
  ./asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml 6458 
  ./asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml 6458 
  ./asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_A_no_direction.xml PRE-CREATION 
  ./asterisk/trunk/tests/channels/SIP/sip_hold/run-test 6458 

Diff: https://reviewboard.asterisk.org/r/4442/diff/



Ashley Sanders

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