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Summary:ASTERISK-11793: lost packets when using Polycom 550 and ulaw codec to outbound sip also using ulaw, if change polycom to alaw no lost packets
Reporter:tim allen (tallen8840)Labels:
Date Opened:2008-04-07 15:38:22Date Closed:2011-06-07 14:07:25
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:rtcp stats showing lots of lost packets and bad voice quality and Polycom stats show same number of lost packets when placing a call from A Polycom 550 to and outbound (myphonecompany.com) SIP "trunk" using the ulaw codec at both ends. No problems when call terminates at Asterisk (my AutoAttendant) or calling another SIP extension Cisco ATA186 etc) or calling from another SIP extension (Cisco ATA) to outbound trunk with ulaw at both ends. So after checking lots of other stuff I changed the Polycom sip entry to alaw, and now it's all better. Please feel free to contact me you can get my phone number from www.integsys.biz. I will provide xml (Polycom), logs whatever from the phone, Asterisk, etc.

Debian Testing quad xenon 700
Polycom 550 SIP 2.2.2.0084



Asterisk 1.4.18.1~dfsg-1 built by pbuilder @ grnetbox on a x86_64 running Linux on 2008-03-18 22:54:26 UTC


****** ADDITIONAL INFORMATION ******

30ish second call from Polycomm 550 through Asterisk via MyphoneCompany.com to my wife's office (ulaw to ulaw)...

 RTP-stats
* Our Receiver:
 SSRC:          971849803
 Received packets: 170
 Lost packets:  0
 Jitter:                0.0008
 Transit:               -0.0004
 RR-count:      1
* Our Sender:
 SSRC:          940095575
 Sent packets:  173
 Lost packets:  0
 Jitter:I>              0
 SR-count:      1
 RTT:           0.000000
 RTP-stats
* Our Receiver:
 SSRC:          1956732908
 Received packets: 173
 Lost packets:  0
 Jitter:                0.0000
 Transit:               0.0001
 RR-count:      1
* Our Sender:
 SSRC:          1792228485
 Sent packets:  170
 Lost packets:  63562
 Jitter:                0
 SR-count:      1
 RTT:           0.004000


same call from Polycomm 550 through Asterisk via MyphoneCompany.com to my wife's office (alaw to ulaw)...

RTP-stats
* Our Receiver:
 SSRC:          64712105
 Received packets: 413
 Lost packets:  0
 Jitter:                0.0006
 Transit:>              -0.0001
 RR-count:      0
* Our Sender:
 SSRC:          512960268
 Sent packets:  413
 Lost packets:  0
 Jitter:I>              0
 SR-count:      1
 RTT:           0.000000
 RTP-stats
* Our Receiver:
 SSRC:          1247948489
 Received packets: 413
 Lost packets:  0
 Jitter:                0.0002
 Transit:               0.0005
 RR-count:      0
* Our Sender:
 SSRC:          1450161644
 Sent packets:  413
 Lost packets:  0
 Jitter:                0
 SR-count:      1
 RTT:           0.001000

Comments:By: jolan (jolan) 2008-06-05 19:11:28

Can you try this?

This seems to workaround the issue for me.

Index: rtp.c
===================================================================
--- rtp.c (revision 120859)
+++ rtp.c (working copy)
@@ -2663,8 +2663,10 @@
    if (rtp->lastts > rtp->lastdigitts)
        rtp->lastdigitts = rtp->lastts;

+ /* XXX polycoms can't cope with high timestamps on the initial packet
    if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO))
        rtp->lastts = f->ts * 8;
+ */
By: Olle Johansson (oej) 2008-07-06 07:45:44

tallen8840: Please test jolans patch.
Looking forward to your feedback.
Thanks.

Jolan: All patches, however small, needs to be uploaded as an attachment so that we can check the license agreement. Please upload this patch. Thank you.
By: Terry Wilson (twilson) 2008-11-17 17:13:15.000-0600

No feedback for over 4 months.  If someone is still interested in this bug, feel free to reopen it.