Comments (16)
Hi,
That's not planned. IAX2 is really different from SIP. I'll try first to make
something suitable in SIP :).
As far as I know, pjsip, the native sip stack on which csipsimple is based,
doesn't
support IAX2.
If the pjsip team decide to add something for IAX2, csipsimple will have this
improvement too.
Another solution would be to integrate another native opensource and portable
stack,
but that's really far to be a priority.
I don't reject this request, but unless somebody else wants to contribute for
that,
it is postponed for a long time.
Original comment by [email protected]
on 31 May 2010 at 8:02
- Added labels: Priority-Low, Type-Enhancement
- Removed labels: Priority-Medium, Type-Defect
from csipsimple.
I vote for this because of its smaller bandwidth requirements and better
reliability
through firewalls. (Stun not required.) Huge advantage for android users on
limited
data plans.
Original comment by [email protected]
on 1 Jun 2010 at 10:59
from csipsimple.
There is a library. Info provided by sipdroid user. Could win some market
share. http://code.google.com/p/sipdroid/issues/detail?id=244
Original comment by [email protected]
on 8 Jun 2010 at 12:32
from csipsimple.
voting for that.
IAX has great potential at mobiles, since it have some improvements in network
part (better NAT handling, among others)
Original comment by zdevel
on 20 Jun 2010 at 4:12
from csipsimple.
vote for iax2 in android pls
as soon as I opened port 5060 to the world I got several dict attacks on sip
login which basically DOS'ed my asterisk server, I had to firewall 5060 to just
a few known IP's that will use my 'roid phone over wifi (will trial fail2ban to
see if that alleviates the attacks)
I use an IAX client on my MacBook (Loudhush or JackenIAX) and have never seen
an attack on 4569 on my asterisk server
also as mentioned IAX2 uses less b/w and avoids NAT issues
apart from this THANKS FOR CSipSimple
WORKS LOVEY ON ZTE BLADE (orange san francisco) under Gingerbread 2.3.3
Cyanogenmod7 (both RC1 and nightlies 1-15 so far)
audio routed correctly to earpiece and then loudspeaker when loudspeaking is
chosen
sometimes audio quality lacks despite using codec 722 - trying KEEP AWAKE
DURING CALL to see if that makes a difference
keep up the great work
Original comment by [email protected]
on 10 Mar 2011 at 8:24
from csipsimple.
try IaxAgent by Johan Cloetens?
http://www.belgianwaves.com/
http://twitter.com/belgianwaves [I am not a twit]
http://www.appbrain.com/app/iaxagent-beta/com.bw.iax.ui
http://pbxinaflash.com/forum/showthread.php?t=7100
IAX handles NAT traversal orders of magnitude better than SIP
Original comment by [email protected]
on 23 Jun 2011 at 11:59
from csipsimple.
IAX is a hack, a toy, a pseudo-protocol (instead of separating signalling and
media as *any* modern VoIP protocol it forces both streams to go together).
IAX does not allow IM neither presence. IAX has several design issues and
critical vulnerabilities and bugs in its main implementation (Asterisk).
IAX was created by Mark Spencer because he does not properly understand SIP
protocol (given the old-fashion-PBX implemented in Asterisk that is not a
surprise for me).
There is enough mechanisms in SIP for handling NAT, in client and server side.
In the other side IAX requires that the media goes always to the central server
which could be so far from both callers. This is a pain and un feasible in many
environments, generating latency and so.
IAX is just supported by a few (very few) phones and just some PBX's (the
anti-cool-DTMF-based Asterisk and FreeSwitch). Must we asume that CSipSimple is
designed to "work with Asterisk"? Please not.
Please forget IAX. Just people with no knowledge of SIP is in favour of IAX
(just because "it seems easier").
Original comment by [email protected]
on 27 Jul 2011 at 1:19
from csipsimple.
Whether you're for or against IAX2, matters not. The fact that SIP and IAX
exsist, is sufficient enough to make them accessable to any user, anywhere.
CSipSimple is a great, stable program and if IAX can be added to it, then I say
go for it.
Will save me running 2 apps and if it can all function in CSipSimple, that
would only be a good thing..
Original comment by [email protected]
on 5 Dec 2011 at 5:33
from csipsimple.
Yep, actually I did a big effort on separating the SIP stack from the rest of
the program. One of the next step will be to allow other sip stacks than pjsip
(for example the stock gingerbread one or doubango, which is another great sip
stack).
If this can be accomplished, I guess that it could be a good starting point to
add other VoIP protocols support. The idea is to allow this kind of third party
voip backend to be loaded as plugins apps, so would be a good compromise.
Android os is very well designed to allow to extends apps features with other
plugins apps.
Well, as for this point, still open, still not in my higher priorities but
contributions are welcome ;).
Original comment by [email protected]
on 5 Dec 2011 at 5:42
from csipsimple.
@Comment 7:
IAX2 does support messaging and IIRC also presence. Combining the signaling and
the audiostream into one connection has advantages in regards to protocol
overhead. Additionally, an encrypted IAX2 stream is significantly harder to
dissect than SIP because of the combination of signaling and audio.
One of the biggest disadvantages of SIP is that it's very hard to operate a
nat'ed PBX and it's impossible to run SIPS/SIPTLS through a nat'ed PBX.
Also, the combination of nat'ed client AND nat'ed PBXs is next to impossible to
realize without dirty hacks which no firewall-admin will go along with.
SIP is fine for LAN or VPN applications. For the open internet and especially
for mobile devices, IAX2 is the better choice.
Only people which need cleartext protocols because they don't understand binary
protocols are against IAX2 ;).
Original comment by [email protected]
on 4 Dec 2012 at 4:08
from csipsimple.
>IAX is a hack, a toy, a pseudo-protocol (instead of separating signalling and
media as *any* modern VoIP protocol it forces both streams to go together).
LoL. First of all IAX2 was designed because of problems with SIP, especially in
NAT environment. For an average user multiplexing works much better then
separate RTP and signalling streams, and BTW, multiplexing is VERY common in
telephony. Also IAX adds some features, including trunking support, dialplan
exchange, etc, but it is a very different story. For me main benefits are:
1) It works via any NAT without ugly hacks like STUN.
2) In case of mulch-channel connection (> 1 stream) it saves bandwidth.
3) Better interoperability - too many ways in SIP to do encryption, pass DTMF,
specification is not clear in many aspects, etc.
IAX2 is supported not only by Asterisk, but also by number of hardware and
software products, including FreeSwitch. Some carriers providing IAX2
connection as well. So, from my point of view it would be great to see IAX2
support in CSIPSIMPLE. And for religious fanatics i would recommend just not to
use it but not start this stupid flame.
Original comment by [email protected]
on 20 Jan 2013 at 7:05
from csipsimple.
Yes , We need IAX2 Support.
SIP is pain for nat and firewalls
When do you think IAX2 version release ?
Thanks
Mohamed
Original comment by [email protected]
on 16 Dec 2013 at 7:51
from csipsimple.
[deleted comment]
from csipsimple.
IAX is only producing less overhead on several calls .. unusual on cellphones.
calls on hold are not using a RTP channel as they are on hold on the Server.
a big disadvantage of IAX2 for mobile .. it doesnt work via TCP, UDP is a
battery killer as allready known due the max refresh time of 60sec on some
cellnetworks even down to 30sec.
In Comparation with TCP 15min ore more, typical is 60 min
i think to implement IAX2 for a Cellphone is not worth.
(as much i like IAX2)
IAX2 does include even the RTP stream, thats maybe one reason why there is
never TCP forseen.
RTP via TCP will be a bandwith killer and using much more Batterie power while
in Call. beside of the usual larger audiobuffer to overcome delay in the stream
caused due retry on IP-Stack level, out of control of IAX2
on the end IAX2 was developed to multiplex streams into one, and share the
channel TDMA like based .. which will never be realiable via TCP
Original comment by [email protected]
on 20 Feb 2014 at 12:14
from csipsimple.
Hi Guys,
Did you create Iax2 for Android ?
Thanks
Original comment by [email protected]
on 30 Apr 2014 at 8:18
from csipsimple.
Presence could be provided via STOX or CUSAX, the two perspectives on Xmpp and
SIP intersections. Jingle (xmpp) might well accomplish the NAT bliss claimed by
IAX proponents who cannot also provide themselves entriprise configurations.
Original comment by [email protected]
on 22 Mar 2015 at 12:44
from csipsimple.
Related Issues (20)
- PSP Wifi mode : Sound becomes choppy/garbled when the phone goes to sleep in mid-call HOT 22
- After SIP call, next GSM call starts in "Loudspeaker" mode HOT 4
- sip outgoing only - result crash when dialing HOT 2
- Ability to put call on HOLD HOT 3
- Register timeout always reverts to 15 sec (Expert account wizard) HOT 15
- Keep-Alive Interval always reverts to blank (0) HOT 1
- Expert wizard: Data Type not initialized with account DB value HOT 12
- pre5 - Cannot disable or change Keep-Alive HOT 8
- Cannot erase Proxy URI field once it has been filled in HOT 2
- Enhancement: Account wizard for Callcentric HOT 5
- Battery consumption HOT 143
- Micro / Speaker booster option HOT 1
- Cannot dial USSD codes while CSipSimple installed HOT 19
- new account HOT 2
- Phone calls do not pause media HOT 30
- No audio / Poor audio on Nexus One HOT 15
- Account Status Labels - More Descriptive and Localizable HOT 3
- Account name not shown on X10 Mini HOT 9
- [Manufacturer driver] Echo on Xperia X10 HOT 16
- Connect to SIP provider only on specific WiFi network
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from csipsimple.