sip calls garbled (PCMu/a)

Bug #578009 reported by Eric Drechsel on 2010-05-09
12
This bug affects 2 people
Affects Status Importance Assigned to Milestone
telepathy-sofiasip
Incomplete
Medium
telepathy-rakia (Ubuntu)
Low
Unassigned

Bug Description

Binary package hint: empathy

I use Gizmo (sipphone.com) with Google Voice and a SIP client on my laptop to place/receive voice calls on the cheap.

Twinkle is my benchmark SIP client, since I've found it to be very reliable compared to other clients available in the Ubuntu repos, especially "on the go" connected to random NATed wireless access points.

I would like to use Telepathy/Farsight/Empathy for SIP calls. In previous releases I had experienced registration errors, but now the registration seems to work properly.

Unfortunately, the audio during calls is garbled (syllables seem to appear out of order, drop etc), while in Twinkle with the exact same setup, audio is solid.

The empathy log and twinkle log both show the codec negotiated as PCMU/A.

I have experienced this in many network environments and on several different computers, and I believe it is a problem for everyone. Currently I am on a campus network which gives me a public IP, so it is not specifically an issue with NAT.

Can other people report their success/failure with SIP calls? Please include the logs (from Help > Debug) for Empathy and sofiasip.

Eric Drechsel (ericdrex) wrote :
Eric Drechsel (ericdrex) wrote :
Eric Drechsel (ericdrex) wrote :

these logs show a session where I start empathy, place a call to "echo" (Gizmo's test service), listen to garbled/dropping audio for a few seconds, and hang up.

Omer Akram (om26er) wrote :

The issue you are reporting is an upstream one and it would be nice if somebody having it could send the bug to the developers of the software http://telepathy.freedesktop.org/wiki/Bugs

affects: empathy (Ubuntu) → telepathy-sofiasip (Ubuntu)
Changed in telepathy-sofiasip (Ubuntu):
importance: Undecided → Low
status: New → Incomplete

I use Gizmo service with Google Voice. Twinkle is my benchmark SIP client, since I've found it to be very reliable compared to other clients available in the Ubuntu repos, especially "on the go" connected to random NATed wireless access points.

I would like to use Telepathy/Farsight/Empathy for SIP calls since it integrates well with the Gnome desktop and unifies my realtime communications. In previous releases SIP registration and call connection were broken, but now these issues are resolved.

Unfortunately, audio during calls is garbled (syllables seem to appear out of order, drop etc), while in Twinkle with the exact same setup, audio is solid. I've tested on a variety of networks, including both fast, public IP connections and behind NAT, with the same results: Twinkle voice quality is solid, Empathy voice quality is unusably glitchy.

The empathy log and twinkle log both show the codec negotiated as PCMU/A.

Since this is a quality issue, not one that is easily traced, I would suggest other interested users test with both Empathy/sofiasip and Twinkle (or other solid client) and post results.

btw, I'm using telepathy-sofiasip 0.6.2-1 on Ubuntu Lucid

downstream bug: https://bugs.launchpad.net/ubuntu/+source/telepathy-sofiasip/+bug/578009

Sounds like an issue for Farsight.

Eric Drechsel (ericdrex) wrote :
Changed in telepathy-sofiasip (Ubuntu):
status: Incomplete → New

Maybe twinkle has a dynamic jitterbuffer (we don't yet) or something like that. Can you make a tcpdump of a call in a situation where twinkle sounds fine and we don't ?

Omer Akram (om26er) wrote :

thanks for sending this bug upstream

Changed in telepathy-sofiasip (Ubuntu):
status: New → Triaged
Changed in telepathy-sofiasip:
importance: Unknown → Medium
status: Unknown → Confirmed
Eric Drechsel (xerdcire) wrote :

Hmm, I just turned off pulseaudio and switched to alsa (Lucid) and the calls are clear now. So this seems to be the fault of pulseaudio (pending further testing)

Eric Drechsel (xerdcire) wrote :

see http://wiki.pdxhub.org/guides/linux_on_laptops for instructions to switch gstreamer to use ALSA

Changed in telepathy-sofiasip:
importance: Medium → Unknown
Changed in telepathy-sofiasip:
importance: Unknown → Medium

I can confirm Erics bug.

My sip works perfectly with ekiga ( not a ekiga.net account), but with empathy/telepathy the sound is "garbled"

Kees van den Broek (kvdb-kvdb) wrote :

Easier instructions for switching to ALSA. It's still needed to prevent garbled sound in Ubuntu 12.04:

$ gstreamer-properties

Default input -> Choose plugin: ALSA, device: Default

Colin Watson (cjwatson) on 2012-10-10
affects: telepathy-sofiasip (Ubuntu) → telepathy-rakia (Ubuntu)

Can someone make a tcpdump when the problem appears ?

Changed in telepathy-sofiasip:
status: Confirmed → Incomplete
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