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Sagemcom 2704N and SIP / UDP ports for RTP : no incoming audio on VOIP with Aste

racquel
Grafter
Posts: 181
Thanks: 4
Registered: ‎21-11-2008

Sagemcom 2704N and SIP / UDP ports for RTP : no incoming audio on VOIP with Aste

To expand on a cryptic subject line, basically, I have an Asterisk VOIP server running on a DigitalOcean VPS, and there is no audio on my Gigaset N300IP phone behind my Sagemcom 2704n router.
I know that the Asterisk server is OK as it works fine via my ISTP (Voipfone), and it works via MicroSIP if I enable "allow IP rewrite" and via Zoiper if I tick "use rport media".
So, following some guides, I limited the ports to a certain range, then opened and forwarded that UDP range on the router, to my Gigaset phone.
Now, here's the thing - I'm sure this didn't happen with my old Technicolor TG582n router.
I also read something about SIPALG causing problems with some routers. Can't find anything to match that setting in my router.
Looking at my Asterisk box for the call, I can see that it is setting an RTP port within the range I have opened on my router, and have set both in my phone and on Asterisk.
To summarise:

  • My Asterisk box can connect to my ITSP (Voipfone).

  • My Gigaset can connect to Voipfone.

  • My Gigaset can connect to Gigaset's own service - echo test works on both

  • I can connect to my Asterisk box via a SIP softphone providing port rewrite/rport is on.

  • However, when connecting my Gigaset phone to my Asterisk box, even with ports open and forwarded, there is no audio.


The problem I am having with narrowing it down is that if it was my VPS, there would be no audio to from it to Voipfone.
If it was my own router, there would be no audio from me to Voipfone.
Totally lost now - I think I've read just about every possible related wiki entry, FAQ and Stackoverflow post about NAT and VOIP and still none the wiser.
And yes, I've google and can find no equivalent to "rport" on my Gigaset - am I completely stuffed in that respect now?
Can anyone shed any light over whether this might be a Plusnet router issue, or should I be looking elsewhere for the problem?
3 REPLIES 3
MisterW
Superuser
Superuser
Posts: 14,575
Thanks: 5,411
Fixes: 385
Registered: ‎30-07-2007

Re: Sagemcom 2704N and SIP / UDP ports for RTP : no incoming audio on VOIP with Aste

Hi, It's almost certainly a NAT issue.
Quote
My Gigaset can connect to Voipfone.
I guess that is using an outbound proxy ? that might explain why it works
I don't have a 2704n but it probably does have  a SIP ALG and it could well be interfering. Also, there are quite a few threads on here relating to problems with port forwarding
on the 2704 so eevn though you think you have forwarded the rtp ports to the N300, it may not actually be doing it. I believe the forwarding works better if the 2704n has allocated the IP via DHCP rather than having a static IP set on the N300, but don't quote me.
There's a thread here about turning off Sip alg https://community.plus.net/forum/index.php?topic=140064.0 so it might be worth dropping Alaric a PM.
On another front, why the change from the TG582n ? 

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racquel
Grafter
Posts: 181
Thanks: 4
Registered: ‎21-11-2008

Re: Sagemcom 2704N and SIP / UDP ports for RTP : no incoming audio on VOIP with Aste

Quote from: MisterW
eevn though you think you have forwarded the rtp ports to the N300, it may not actually be doing it. I believe the forwarding works better if the 2704n has allocated the IP via DHCP rather than having a static IP set on the N300

Hmmm.. I did a "port open test" and it seems to be forwarding the SIP 5060 port at least.
Also noticed under http://192.168.1.254/expert_user.html that there is a setting for port triggering, but having googled it, I'm not sure that's what I want. As I understand it, VOIP opens two sequential ports, so if it triggered say 12345 to listen, but then outgoing speech was on 12346, that that still wouldn't help. Or have I misunderstood that?
Quote from: MisterW
it might be worth dropping Alaric a PM.

Done! Thanks - waiting reply with hope!
Quote from: MisterW
On another front, why the change from the TG582n ?   

Because, having taken 3 months to track down, isolate, repeat and document a bug which caused the router to crash when a SYN packet was sent from Google Chrome while using IPv6 tunnelling (edge case!), I then contacted Plusnet, Thomson, Technicolor - nothing. Zip. Zilch.
So they sent me a new one instead, only for me to find that it can't do tunnelling. Besides, that was back when I thought IPv6 was the future and it would be a good thing to learn. As Plusnet are showing, there's no rush at all Smiley
So the TG582n found a new home as a router/wireless repeater and the 5704N became the new hub.
Threads related to TG582n crashes....
http://community.plus.net/forum/index.php/topic,106578.msg1182706.html#msg1182706
https://community.plus.net/forum/index.php/topic,133622.msg1180076.html#msg1180076
Thanks!
MisterW
Superuser
Superuser
Posts: 14,575
Thanks: 5,411
Fixes: 385
Registered: ‎30-07-2007

Re: Sagemcom 2704N and SIP / UDP ports for RTP : no incoming audio on VOIP with Aste

Quote
As I understand it, VOIP opens two sequential ports,
SIP uses port 5060 for signalling, this isn't usually too much of a problem with NAT since it can usually be kept open using either an outbound proxy or 'keep-alive' traffic which keeps a NAT pinhole open on the router. It's the rtp ports that give the problem, these are allocated by each end dynamically during the call setup, they are typically in the 10000 to 20000 range but the range can be usually be set on the device.
Have you got the nat option set in the Asterisk config for the N300 peer ? I would suggest trying nat=yes & qualify=yes for starters. If that doesn't work then try alternative nat settings of force_rport or comedia
You also probably need STUN configured on the N300 .
Quote
Because, having taken 3 months to track down, isolate, repeat and document a bug which caused the router to crash when a SYN packet was sent from Google Chrome while using IPv6 tunnelling (edge case!), I then contacted Plusnet, Thomson, Technicolor - nothing. Zip. Zilch.
Ah! that explains why the TG582 isn't used anymore. TBH from what I read on here the 2704 is a pretty rubbish alternative though...

Superusers are not staff, but they do have a direct line of communication into the business in order to raise issues, concerns and feedback from the community.