[asterisk-dev] [Code Review] 4286: rtp_engine: avoid payload types above 127
Olle E Johansson
reviewboard at asterisk.org
Fri Dec 19 14:39:54 CST 2014
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You may want to consider fixing RFC 5761 restrictions while you are in there... This prepares for RTP/RTCP muxing in WebRTC
"Given these constraints, it is RECOMMENDED to follow the guidelines
in the RTP/AVP profile [7] for the choice of RTP payload type values,
with the additional restriction that payload type values in the range
64-95 MUST NOT be used. Specifically, dynamic RTP payload types
SHOULD be chosen in the range 96-127 where possible. Values below 64
MAY be used if that is insufficient, in which case it is RECOMMENDED
that payload type numbers that are not statically assigned by [7] be
used first."
- Olle E Johansson
On Dec. 19, 2014, 9:24 p.m., Scott Griepentrog wrote:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4286/
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>
> (Updated Dec. 19, 2014, 9:24 p.m.)
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>
> Review request for Asterisk Developers.
>
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> Bugs: ASTERISK-24367
> https://issues.asterisk.org/jira/browse/ASTERISK-24367
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> Repository: Asterisk
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> Description
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> Valid payload type codes are between 0 and 127 to allow for being stored in 7 bits. During call setup, pjsip validates the SDP and will assert if it encounters an invalid payload type code (see pjmedia_sdp_validate() in pjmedia/src/pjmedia/sdp.c). This assert will be hit if a call is placed to a pjsip endpoint with allow=all set.
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> To avoid this, the previous use 128 for the slin192 format has been changed to 95.
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> Diffs
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> /trunk/main/rtp_engine.c 429845
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> Diff: https://reviewboard.asterisk.org/r/4286/diff/
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> Testing
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> Tested with pjsip calls to allow=all configured extensions.
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> Thanks,
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> Scott Griepentrog
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>
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