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whats_up_skip 2009-12-21 09:27

Setting codec in SIP client
 
Is it possible to set the codec used in the N900 SIP VoIP client?

Andre Klapper 2009-12-21 12:50

Re: Setting codec in SIP client
 
Examples for codecs that you want to set, please?

whats_up_skip 2009-12-21 21:54

Re: Setting codec in SIP client
 
I am not sure which codecs are available, but g729, gsm or ilbc, as these provide better performance over HSPA.

onestop 2009-12-23 04:47

Re: Setting codec in SIP client
 
Quote:

Originally Posted by whats_up_skip (Post 436176)
Is it possible to set the codec used in the N900 SIP VoIP client?

I haven't found anyway to configure the codec using the "VOIP and IM accounts" panel under the settings utility.

Moreover, although I can initiate and receive calls, there is a problem with the outbound voice when I'm using my VoIP provider (normal telephone calls work just fine - it's not a dead mic).

Croc 2009-12-23 09:46

Re: Setting codec in SIP client
 
i can't see any way to change codecs as well, i hope this doesn't mean it defaults to g711 as it would be bad idea for 3g or even on wifi if user got slow upload speed.
for E series there is sip tool on nokia website that let us adjust this so i hope there is a way to change codecs and codecs priority on n900 :(
is quite odd that such simple feature yet so much needed one is missing

blubbi 2009-12-23 12:06

Re: Setting codec in SIP client
 
I agree!

I was searching for the codec too, but no hit jet.

Would be great if you could tie a codec priority list to the Internet connection used (Wifi, 3G, 2,5G, 2G) or autodetection based on currently available bandwidth.

Cheers
Bjoern

whats_up_skip 2009-12-24 07:33

Re: Setting codec in SIP client
 
I suspect by the poor quality of outbound sound on HSPA that it is either defaulting to ulaw (g711) or has the priority set as ulaw first.

I will try to do some more testing tomorrow to see what I can work out.

There is another issue with the VoIP system I noticed:
If you have more than SIP account registered and you select a name from the contact list it offers "phone" or "sip audio" numbers, but it does not tell you which sip account it is going to use.

R-R 2009-12-24 07:39

Re: Setting codec in SIP client
 
Quote:

Originally Posted by onestop (Post 438586)
I haven't found anyway to configure the codec using the "VOIP and IM accounts" panel under the settings utility.

Moreover, although I can initiate and receive calls, there is a problem with the outbound voice when I'm using my VoIP provider (normal telephone calls work just fine - it's not a dead mic).

Same problem here in asterisk 1.6.2.0 with this (sip.conf) :

allow=gsm,ulaw,alaw,h263,g729

I hope this get fixed in the upcoming release as this is a MAIN feature of this phone!

whats_up_skip 2009-12-24 07:46

Re: Setting codec in SIP client
 
Quote:

Originally Posted by R-R (Post 440055)
allow=gsm,ulaw,alaw,h263,g729

What happens if you change it to either:
allow=g729

or

allow=gsm

This might let us know which codecs are installed.

blubbi 2009-12-24 09:03

Re: Setting codec in SIP client
 
Quote:

Originally Posted by whats_up_skip (Post 440050)

There is another issue with the VoIP system I noticed:
If you have more than SIP account registered and you select a name from the contact list it offers "phone" or "sip audio" numbers, but it does not tell you which sip account it is going to use.

See the bug in my signature. Please vote for it. It might as well resolve your issue.

Cheers Bjoern

tom4_u 2009-12-24 09:44

Re: Setting codec in SIP client
 
hi Guys ,
will some one help me out here to setup my SIP account

i have an username say :tom@voip.sip.com
and password say :123
and my sip server is say voip.sip.com
and proxy server is the same i.e voip.sip.com

Now in the n900 sip account menu

i have filled it out like this

username : - tom@voip.sip.com
password :- 123

[Advanced settings]

username : ?
Outbound proxy: ?
discover public address is checked
loose routing is unchecked
keep alive mechanism set auto
kep alive period set to auto
Auto detect STUN checked


with the above settings i m not able to sign in.

I have also tried varios combinations like
username : tom
password : 123
[advanced settigs

username : tom
outbound proxy : voip.sip.com
rest all r the same

I use the same setting in my nimbuzz cleint(PC) and i m able to make calls
What is that i m doing wrong here?

Thank you
TOM

rmarcus 2009-12-24 09:47

Re: Setting codec in SIP client
 
Quote:

Originally Posted by blubbi (Post 440103)
See the bug in my signature. Please vote for it. It might as well resolve your issue.

Cheers Bjoern

In mine it displays separate contact numbers inside the contact so I can choose which SIP use, either skype, or the other one I use.

:)

blubbi 2009-12-24 10:00

Re: Setting codec in SIP client
 
Mhh, I might figure out the used codec.

If I call myself my FritzBox 7270 can show details about the call.
I am not sure if it reveals the used codec, but it is worth a try.

I'll check that as soon as I am at home (might be next year)

Cheers
Bjoern

R-R 2009-12-24 14:22

Re: Setting codec in SIP client
 
Quote:

Originally Posted by whats_up_skip (Post 440064)
What happens if you change it to either:
allow=g729

or

allow=gsm

This might let us know which codecs are installed.

I will have to test this soon, i think i had to disable g711 at first as it would even call with that one at the beginning of the list.
(which is apparently bad over wifi, so not an issue.)

check out bug 7302!


Quote:

Originally Posted by rmarcus (Post 440121)
In mine it displays separate contact numbers inside the contact so I can choose which SIP use, either skype, or the other one I use.

:)

Same here but that's not the problem... when you receive a call through a real SIP provider it won't show up as a call from "john smith" in the call log but from 5551235551@12.13.14.15 sip etc...

please vote! :-)

Also please vote on these related issues:

Direct (no proxy or registrar) SIP URI dialing

SIP Voice Mail message notification

whats_up_skip 2009-12-24 21:12

Re: Setting codec in SIP client
 
Quote:

Originally Posted by tom4_u (Post 440118)
hi Guys ,
will some one help me out here to setup my SIP account

TOM

Try the following:
Pennytel

User name 888917XXXX@sip.pennytel.com (this uses your Pennytel number)
Password zXXXXXX3 (Pennytel supply you with a password when you join)
Then go to Advanced settings
Use for telephone numbers Make sure this box is checked
User name 888917XXXX (Your Pennytel number)
Transport Auto
Outbound proxy sip.pennytel.com
Port 5060
Discover public addresses Make sure this box IS checked
Loose routing Make sure this box is NOT checked
Keep-alive mechanism Auto
Keep-alive period Auto
Auto detect STUN Make sure this box IS checked

whats_up_skip 2009-12-27 08:03

Re: Setting codec in SIP client
 
Ok, I have been doing some testing and when the N900 is only given the option of connecting with g729 it connects and works. This means at least there is g729 codec built in. I haven't had a chance to test this much further yet.

wierdo 2009-12-27 08:19

Re: Setting codec in SIP client
 
N900 follows the server's codec choice. I have mine set to g729,ulaw,gsm on the server side and in practice the choice is g729.

That's not to diminish the usefulness of forcing a different codec order on the client side, though!

Croc 2009-12-27 16:47

Re: Setting codec in SIP client
 
Quote:

Originally Posted by whats_up_skip (Post 442746)
Ok, I have been doing some testing and when the N900 is only given the option of connecting with g729 it connects and works. This means at least there is g729 codec built in. I haven't had a chance to test this much further yet.

and you did that how?
can we change codec options with pennytel?

whats_up_skip 2009-12-27 20:34

Re: Setting codec in SIP client
 
I made this changes on my Asterisk box. I don't think Pennytel gives you this option and this is the reason why we still need the ability to set the codec on the N900.

tom4_u 2009-12-28 06:37

Re: Setting codec in SIP client
 
Quote:

Originally Posted by whats_up_skip (Post 440768)
Try the following:
Pennytel

User name 888917XXXX@sip.pennytel.com (this uses your Pennytel number)
Password zXXXXXX3 (Pennytel supply you with a password when you join)
Then go to Advanced settings
Use for telephone numbers Make sure this box is checked
User name 888917XXXX (Your Pennytel number)
Transport Auto
Outbound proxy sip.pennytel.com
Port 5060
Discover public addresses Make sure this box IS checked
Loose routing Make sure this box is NOT checked
Keep-alive mechanism Auto
Keep-alive period Auto
Auto detect STUN Make sure this box IS checked


pennytel.com is blocked by ISP in this part of the country, and the voip server i use is sip.voiparound.com, it works fine woth my Nimbuzz and talkonaut clinet, the PC version and s60 v5 version, but cant understand y its not working with n900. Is it because i m doing something wrong with the config settings?

Thanks

ccc1 2009-12-28 08:31

Re: Setting codec in SIP client
 
Quote:

Originally Posted by tom4_u (Post 440118)
hi Guys ,
will some one help me out here to setup my SIP account [...]

and how this is releated to "Setting codec in SIP client"? it might be a better idea if you start a new thread ...

Croc 2010-01-10 01:22

Re: Setting codec in SIP client
 
was there any discoveries on this topic? on wlan it is ok with my adsl but not being able to select other codecs while on 3G makes sip client useless :(

baergaj 2010-01-13 18:14

Re: Setting codec in SIP client
 
Quote:

Originally Posted by Croc (Post 461871)
was there any discoveries on this topic? on wlan it is ok with my adsl but not being able to select other codecs while on 3G makes sip client useless :(

You can configure the codecs for sip by editing /etc/stream-engine/gstcodecs.conf

You can disable a codec by putting "id=-1" in its configuration block.

Obviously, you need to sudo gainroot to edit this.

Croc 2010-01-14 01:14

Re: Setting codec in SIP client
 
Quote:

Originally Posted by baergaj (Post 468837)
You can configure the codecs for sip by editing /etc/stream-engine/gstcodecs.conf

You can disable a codec by putting "id=-1" in its configuration block.

Obviously, you need to sudo gainroot to edit this.

thanks a milion for this info :)

arkanoid 2010-01-14 01:21

Re: Setting codec in SIP client
 
Quote:

Originally Posted by baergaj (Post 468837)
You can configure the codecs for sip by editing /etc/stream-engine/gstcodecs.conf

You can disable a codec by putting "id=-1" in its configuration block.

Obviously, you need to sudo gainroot to edit this.

Wow, what happens if you enable *video* codecs as well? ;-)

kodomo 2010-01-22 14:11

Re: Setting codec in SIP client
 
Quote:

Originally Posted by baergaj (Post 468837)
You can configure the codecs for sip by editing /etc/stream-engine/gstcodecs.conf

You can disable a codec by putting "id=-1" in its configuration block.

Obviously, you need to sudo gainroot to edit this.

Hm - for some reason, the sip client does not include GSM as an option upon codec negotiation (although there's no id=-1 below it). Is there an additional file to edit?

basically, I've got:
a=rtpmap:18 G729/8000
a=rtpmap:97 ILBC/8000
a=fmtp:97 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:104 speex/8000
a=rtpmap:100 DV/90000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 events=0-15
a=rtpmap:102 telephone-event/90000
a=fmtp:102 events=0-15


Although:
[audio/GSM]
clock-rate=8000

[audio/G729]
clock-rate=8000

...

sky4vip 2010-06-10 18:46

Re: Setting codec in SIP client
 
I'm using gizmo5, and have found that it will ONLY connect using pcmu (the high bandwidth solution). This makes callers complain of my voice 'skipping' if I'm near a place that has sketchy 3g, or goes 3g-3.5g (using tmobile US network). I've tried editing id=-1 in .conf file above, and have found w/ the other codecs, the phone appears to answer an incoming call on my end, but just drops it immediately. Callers on the other end report never connecting and being sent to voicemail or the ilk (as if I never picked up, despite what the phone says.

Gizmo5 states that they support ilbc & the low(er) bandwidth PCM, as well as GSM. None of them seem to be working.

Is anyone else using Gizmo and seeing similar results? Is there something else I should be doing. I have the suggested SIP setup from gizmo's end, and am using pr1.2

NOTE. This setup WORKS, it just uses a (very) high bandwidth, unsuitable for anything but (very) good 3g connections.

GTalk and Skype work well; I would switch google voice forwaqrding to them, but prefer the way that the contact's info gets picked up on SIP; plus, right now, gizmo5 is free.

travla 2010-07-22 00:27

Re: Setting codec in SIP client
 
Quote:

Originally Posted by whats_up_skip (Post 440768)
Try the following:
Pennytel

User name 888917XXXX@sip.pennytel.com (this uses your Pennytel number)
Password zXXXXXX3 (Pennytel supply you with a password when you join)
Then go to Advanced settings
Use for telephone numbers Make sure this box is checked
User name 888917XXXX (Your Pennytel number)
Transport Auto
Outbound proxy sip.pennytel.com
Port 5060
Discover public addresses Make sure this box IS checked
Loose routing Make sure this box is NOT checked
Keep-alive mechanism Auto
Keep-alive period Auto
Auto detect STUN Make sure this box IS checked


Thanks whats_up_skip, settings worked a treat. Just one note for other PennyTel users, the reference to the first user name above (888917XXXX@sip.pennytel.com) is actually the address in the SIP settings.

Regards,

travla

ilf 2011-09-04 02:41

Re: Setting codec in SIP client
 
I'm sorry to resurrect this topic but I'm trying to set a specific codec in the SIP client i.e. the GSM codec. Has anyone been able to do that? It seems from the info I gathered as around the forum, that gsm codec is not working in the SIP client of N900.

I'm trying to do a SIP conversation between my N900 and my GF's Ginger-breaded Hero and of course the piece of **** that Android is, it does support only gsm and g.711. The problem is that when trying to connect to the N900 with g.711 Android's phone stack dies a horrible death, not to mine displeasure, I have to admit. At the same time N900 doesn't seem to work with the gsm codec at all in SIP mode.

I'm running my own SIP (kamailio) server (not media server i.e. asterisk/freeswitch, but real SIP server) so essentially I don't want to rewrite the SDP and kill the idea behind SIP by forcing something to the phones that they don't want to do.

By the way it seems my E71 is quite happy with the PCM (g.711) of the Android and vice-versa, so the stupid, created around 5 pm on a Friday before a long-weekend, OS doesn't seem to die, so if you are willing to share some g.711 magic for the N900, it would be great. Also if someone knows a way to enable speex or iLBC on the Android OS through the native SIP client, that would really make my day.

xes 2015-04-18 17:00

Re: Setting codec in SIP client
 
@ilf

1 - check your wifi settings, maybe disabling powersave could help (on both android and N900)
2 - install a reliable sip client into android (Csipsimple works great)
3 - if you have a low bandwidth or high latency connection use gsm(8k) / g729 _NOT_ g711a/b
4 - if you know what you are doing, check your sip server nat and sip connection tracking (and/or stun server setting)
5 - check also your provider about complaints of someone else about QOS on sip calls (today it's a very common behavior to limit voip).

6 ..good luck!


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