[asterisk-dev] [Code Review] 3687: media format improvements: Update packetization handling; improve rtp_engine's ast_rtp_codecs handling

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3687/ --- Review request for Asterisk Developers. Repository: Asterisk Description

[asterisk-dev] Proposed removal of deprecated modules in Asterisk 13 (cdr_sqlite, chan_gtalk, chan_jingle, res_jabber, chan_h323, app_readfile, app_dahdibarge, app_setcallerid, app_saycountpl)

The following modules are currently deprecated in trunk. This e-mail is a proposal to remove them from trunk in preparation for Asterisk 13. All of these modules have been deprecated for at least two release cycles (the one they were deprecated in initially, +1 additional release). Rationale for th

Re: [asterisk-dev] [Code Review] 3679: WebRTC: Add SHA-256 support, change DTLS-SRTP negotiation, add finer grain control of things.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3679/ --- (Updated June 28, 2014, 9:51 p.m.) Review request for Asterisk Developers.

Re: [asterisk-dev] [Code Review] 3679: WebRTC: Add SHA-256 support, change DTLS-SRTP negotiation, add finer grain control of things.

> On June 28, 2014, 8:35 p.m., Matt Jordan wrote: > > /branches/11/res/res_rtp_asterisk.c, lines 1061-1079 > > > > > > Does this leak read_BIO and write_BIO on the rtp/rtcp structs? > On June 28, 2014, 8:35 p.m.,

Re: [asterisk-dev] [Code Review] 3647: Enable Forward Secrecy (PFS) in TLS

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3647/#review12381 --- Ship it! I'll relent on my desire for better error messages. (

Re: [asterisk-dev] [Code Review] 3653: chan_sip: (Optionally) poll even on first part of TLS message

> On June 27, 2014, 9:42 a.m., Matt Jordan wrote: > > > > Alexander Traud wrote: > > hence a 'retry once' poll may not be sufficient regardless to read all > of the data from the socket. > > I am not sure, I understand you guys. Just to clarify my intentions: The > proposed patch

[asterisk-dev] [Code Review] 3686: WebRTC: Add SHA-256 support to chan_pjsip and add option to make it answer using the offer media transport.

--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3686/ --- Review request for Asterisk Developers. Repository: Asterisk Description

Re: [asterisk-dev] [Code Review] 3658: SIP/SDP fmtp line may include whitespace between attributes

> On June 24, 2014, 7:58 p.m., opticron wrote: > > /branches/11/channels/chan_sip.c, line 11201 > > > > > > It appears that this will need to be merged into 12/trunk when this > > patch goes in. This patch was t

Re: [asterisk-dev] [Code Review] 3653: chan_sip: (Optionally) poll even on first part of TLS message

> On June 27, 2014, 2:42 p.m., Matt Jordan wrote: > > > hence a 'retry once' poll may not be sufficient regardless to read all of the > data from the socket. I am not sure, I understand you guys. Just to clarify my intentions: The proposed patch is not about to fix/resolve all issues in TLS r