| Summary: | ASTERISK-20835: RTP not modified when UAS responds with an OK(200) with other ptime then 20ms | ||||
| Reporter: | not here (looserouting) | Labels: | |||
| Date Opened: | 2012-12-21 05:14:48.000-0600 | Date Closed: | 2013-01-14 16:16:01.000-0600 | ||
| Priority: | Minor | Regression? | Yes | ||
| Status: | Closed/Complete | Components: | |||
| Versions: | 10.11.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Attachments: | ( 0) debug ( 1) extensions.conf ( 2) sip_and_rtp.pcap ( 3) sip.conf | |||
| Description: | my Asterisk 10 svn recievs an INVITE from A with ptime:20 and invites B also with ptime:20.
now B sends an OK with ptime:30. The OK from Asterisk to A contains ptime:20. But now the RTP stream from asterisk to A is the same as from B to asterisk. That means asterisk doesn't change the stream ( 20 40 20 40 20 40 )and is transmitting with a ptime of 30ms (30 30 30 30 30 ). regarding the RFC this shouldn't be an issiue since RFC4566 says:" It should not be necessary to know ptime to decode RTP or vat audio, and it is intended as a recommendation for the encoding/packetisation of audio." but this can lead to strange behaviour of A like synchronisations problems. | ||||
| Comments: | By: Michael L. Young (elguero) 2012-12-21 16:12:34.670-0600 We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Also, a pcap would be helpful as well. By: not here (looserouting) 2012-12-23 10:38:29.405-0600 the debug log is a bit long. It contains the asterisk start and the test call. Nothing else. the pcap has all the related traffic. All magic happened on the same machine. "A" is a softphone (sflpohne). running sip on 35060 "B" is SIPp. I used it to reproduce the Szenario. It first happend with an AVM Fritzbox. 'cause It uses always ptime 30. here running on port 15060 Asterisk itselft is running on port 25060 you' ll also see all 4 streams in the pcap A to asterisk asterisk to A asterisk to B B to asterisk By: Matt Jordan (mjordan) 2013-01-14 16:16:01.231-0600 Closing out as a duplicate of ASTERISK-20650. | ||||