| Summary: | ASTERISK-11958: Changing of RTP SSRC between early-session and actual call session | ||
| Reporter: | Francesco Castellano (fcastellano) | Labels: | |
| Date Opened: | 2008-05-02 05:52:41 | Date Closed: | 2011-06-07 14:08:17 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Core/RTP |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Upgrading Asterisk from 1.4.17 to 1.4.19 I noticed some SIP clients (for example Siemens Gigaset C450IP), experienced a silence period (4/5 secs) after the remote party (on PSTN) answered. Looking at the RTP packets, I concluded this is relate to the change of SSRC between the early session established with 183 Session Progress, and the actual call session following the 200 Ok message. ****** ADDITIONAL INFORMATION ****** This seems to be related to issue 0012353, because a similar silence associated to the change of SSRC happens during trasmissions of DTMF. Nonetheless, I cannot understand if such a change is "correct" (standard-compliant) or not. Moreover I was wondering if the removing rtp->ssrc = ast_random(); from ast_rtp_new_source function is a viable solution, even if it is not being included in the release 1.4.19.1 | ||
| Comments: | By: Joshua C. Colp (jcolp) 2008-05-05 09:04:17 This has already been changed. | ||